similar to: Ogg Vorbis - low delay <50ms ?

Salve, one year ago here was already a discussion about a version of Ogg Vorbis, with low delay enough for VoIP http://www.xiph.org/archives/vorbis/200108/0106.html >all transform codecs. They require a fairly large block of samples, i.e. in >Vorbis you need to input 3072 samples (or is it 4096) from each channel >before you can get any output. This could be reduced, trading off

Hello all, I have samba 3 with a share named shareA using these settings: [shareA] comment = Directory Amministrazione path = /col/shareA browseable = no valid users = @amm force group = amm public = no writable = yes create mask = 0770 directory mask = 0770 printable = no where user1, user2, user3 and user4 belong to "amm" group. I would like to

Hi list, I'm trying to use an AudioCodec Mp114, 4 FXO Media gateway. I went trough what i could find in wiki and also trixbox forum and so far no good results. i had this in trixbox frorum : http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup any successful installation? or how to? -------------- next part -------------- An HTML attachment was

On Monday 01 December 2014 03:41 PM, Teemu Huovila wrote: > On 11/30/2014 05:53 AM, AMM wrote: >> __strspn_sse42 (in /lib64/libc-2.14.90.so) > Is it possible that you are encountering this issue? https://bugs.kde.org/show_bug.cgi?id=270925 > Either way, the error seems to stem from your libc implementation (if it is not the valgrind bug). > > If possible, upgrade your

Hi, Another newbie question. I'm trying to figure out how to fit vorbis into the Apple AudioCodec API and it asks your codec to be able to answer questions about itself. One I'm not sure about, does vorbis encode a fixed number of frames per packet for any given track? I know it's VBR, but that doesn't necessarily imply variable # of frames per packet. Thanks, -n8 --

Salve, Imagine ogg vorbis is used to produce radio with free software. A journalist would produce a report end send it with 24kBit/s out from a cricis place somewhere in the world. DRM is going to use MP4 - so his report has to be reconverted with loosing quality :-( Can you imagine to have an free codec someday that would work in embedded radio-reciver like MP4? If yes, should DRM not be open

From: Nathaniel Gray <n8gray@gmail.com> Date: Sep 14, 2005 11:30 AM Subject: Newbie q: decoupling vorbis from ogg To: vorbis-dev@lists.xiph.org Hi, Sorry if this is a newbie question. I'm trying to write an OS X AudioCodec for Vorbis using libvorbis. I'm confused about the libvorbis dependency on libogg. I thought the vorbis spec didn't require ogg as the container, but the

I wanna retranslate some radiostations, but they are running Shoutcast and broadcasting in AAC+ format. Can IceCast relay sound in this format from this server?

Hello, I am currently using Dovecot 2.2.10 on Fedora 16 - 64 bit system I had made v2.2.10 Fedora 16 rpm file using spec file from ATrpms. http://dl.atrpms.net/all/dovecot.spec It has been working well from 6 months or so. Today I tried to make v2.2.15 rpm using same spec file. But "make check" is giving following error: fatal_printf_format_fix

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

unkn0wn wrote: > I wanna retranslate some radiostations, but they are running Shoutcast > and broadcasting in AAC+ format. Can IceCast relay sound in this > format from this server? yes, but any transcoding to any other format/bitrate would have to be done via an external application. karl.

Hi all. Here is my first patch for icecast: auth_hash.diff - this patch created to generate unique URL's that gives access to listeners. It also privides GeoIP checking for radiostations, that have some restrictions to audience from other counties. Here is sample config: <mount> <mount-name>/live</mount-name> <authentication

Added logs and screenshots with screen that appears during before a reboot and errors are reported to AMM in bladecenter -------------- next part -------------- A non-text attachment was scrubbed... Name: xen errors.zip Type: application/zip Size: 194637 bytes Desc: not available URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20140905/63a7d8df/attachment-0001.zip>

On 11/30/2014 05:53 AM, AMM wrote: > __strspn_sse42 (in /lib64/libc-2.14.90.so) Is it possible that you are encountering this issue? https://bugs.kde.org/show_bug.cgi?id=270925 Either way, the error seems to stem from your libc implementation (if it is not the valgrind bug). If possible, upgrade your valgrind, libc etc. br, Teemu Huovila

On 12/01/2014 12:41 PM, AMM wrote: > > On Monday 01 December 2014 03:41 PM, Teemu Huovila wrote: > But Dovecot 2.2.10 (and earlier versions) were not throwing this error. This test was added in Dovecot version 2.2.14. It is also the only reference to strspn() in the whole project. > Can I can ignore it by NOT doing "make check"? I would say you can safely ignore it, but I

Hi all. I have samba 3.0.8 running on a redhat 7.3 server, I can join a domain using rpc, but if I use ads then I get ------------------------------- [root@x2scan root]# net ads join Computers -W SOFTCOM -U addtodomain addtodomain's password: [2004/11/19 14:38:07, 0] utils/net_ads.c:ads_startup(186) ads_connect: No such file or directory [root@x2scan root]#

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

Package: xen-hypervisor-4.4-amd64 Version: 4.4.0-4 Severity: normal -- System Information: Debian Release: jessie/sid APT prefers testing-updates APT policy: (500, 'testing-updates'), (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 3.14-2-amd64 (SMP w/4 CPU cores) Locale: LANG=C, LC_CTYPE=C (charmap=ANSI_X3.4-1968) Shell: /bin/sh linked to /bin/dash

Salve Salve Galera. Tenho a seguinte situacao: Uma placa MD3200 ligada em uma linha telefonica comum(PTSN) e funcionando "belezinha"... Tenho configurado um URA, onde ele atende a ligacao que chegou no canal e solicita o numero do ramal de destino da liga??o: Acontece que ao discar o ramal de destino, ele nao encaminha a ligacao, ficando mudo e posteriormente caindo a liga?ao. Fiz