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Hi, I have a Dell T710 with 4 X 10G ethernet interfaces (2 X Dual port Intel 82599 cards). It is running FreeBSD RELENG_8 last updated on July 13. What I see is packet loss (0 - 40%) on IPv6 packets in vlans, when the machine is not the originator of the packets. Let me try to describe a little more. If a neigbouring machine ping6 it, there will be packet loss. If it act as a router for ipv6,

A related question: I would like to build a CD-Rom server based on Samba where it is possible to swap CDs without mounting them by hand. The volume label should appear automatically in Windows. Is this possible ? -Erwin [John D. Hays - Writes] Erwin, I do this on my system. I have two each 7-disk changers on my Linux box (attached using SCSI) with the appropriate driver configuration. All

There was an interruption in mail service and your memo was not sent. Please re-send your message. Thank-you -----Original Message----- From: samba@samba.anu.edu.au <samba@samba.anu.edu.au> To: Multiple recipients of list <samba@samba.anu.edu.au> Date: Sunday, November 01, 1998 1:24 AM Subject: SAMBA digest 1861 > SAMBA Digest 1861 > >For information on unsubscribing

...it out, encodes it, and transmits it. Otherwise it goes back to sleep. I was going to change this to use notifications but haven't yet since it's been working so well... Are you saying that notifications aren't implemented for DirectSoundCapture buffers? Tom <p>John Hayes (jhayes@thereinc.com) wrote: > > Right - and I deal with that on the receiver end based on an approximation > of sender's and receiver's responsiveness - the minimum latency I've been > able to get into the system is about 150 ms. Of that, jitter buffering is > about 40-100ms. I...

Hi, After FreeBSD changed from using -O2 to using -O on their ARM port, I found that sshd stopped working. (gcc version 4.2.1 20070719 [FreeBSD]) I have downloaded openssh-SNAP-20080220.tar.gz and the code still look the same. Anyway looking into it, I found that the problem is in monitor_fdpass.c in the functions mm_send_fd and mm_receive_fd. Using -O2 used to align the tmp array on a 4 byte

Well, colleagues, I'm stuck a bit. I tried many different ways to setup system with vinum root (the only reference I found yet besides old "bootstrapping vinum" article is Joerg's commit message: http://freebsd.rambler.ru/bsdmail/cvs-all_2003/msg01225.html I failed. I have 4-stable system set up at ad0, and tried to set up pair of drives for new system at ad2 and ad3 (actually,

CVSup is slow, insecure, and a memory hog. However, until now it's been the only option for keeping an up-to-date ports tree, and (thanks to all of the recent work on vuxml and portaudit) it has become quite obvious that keeping an up-to-date ports tree is very important. To provide a secure, lightweight, and fast alternative to CVSup, I've written portsnap. As the name suggests, this

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

The audio frame speex generates sounds pretty terrible most of the time, and I don't use it for jitter correction instead I just use it for dropped packets - so I usually drop the late packet. It sounds acceptable as long as I drop less than 5% of traffic (dropping 2 in a row makes a bad robot noise, so I reset the stream in that case). The good news is that on an unsaturated DSL line jitter

If I encode something in ultra-wideband, can I decode it in wideband by chopping off bytes in every frame? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed.

Right - and I deal with that on the receiver end based on an approximation of sender's and receiver's responsiveness - the minimum latency I've been able to get into the system is about 150 ms. Of that, jitter buffering is about 40-100ms. I'd love to figure out how to get that down without killing myself on thread switching or Win32 kernel calls, but ms has to actually implement

I've been playing with speex for use in a VoIP application between PC's. One thing I've found (correlating to the documentation) it that speex runs much faster and produced much better output when it's fed a 32kHz signal instead of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the

I'm trying to get speex to encode a bit faster, mainly by rewriting a few functions in SSE and translating the GCC __asm__ to VC __asm. There's 2 functions I'm targeting, first is vq_nbest which consumes 40% of the time at high complexity and split_cb_search_shape_sign. Which consumes just over 30%. I've split out two functions from: cb_search_precompute_energy - loop at the