search for: 200kbps

...egress HTB (because ipables marks packages based on --mac-source ). So my only choice is using ingress or u32. So this is how I did it: I called bellow script add_shaping DEV="eth0" tc qdisc add dev $DEV root handle 1: htb default 20 tc class add dev $DEV parent 1: classid 1:1 htb rate 200kbps ceil 200kbps tc class add dev $DEV parent 1:1 classid 1:15 htb rate 10kbps ceil 13kbps prio 3 tc class add dev $DEV parent 1:1 classid 1:20 htb rate 150kbps ceil 187kbps prio 2 tc qdisc add dev $DEV parent 1:15 handle 150: sfq perturb 10 tc qdisc add dev $DEV parent 1:20 handle 200: sfq perturb 10...

Hi, I have some files size 400MB+ that my rsync ends on, error message is : "Write failed: Broken pipe". Without this "big files" everything works smoothe. I searched to find anything about this problem but so far without any luck. Source machine : Solaris 8, rsync-2.4.6 Destination machine : FreeBSD 4.1.1, rsync 2.4.6 Command run(from source machine): rsync

...curious if anyone was able to find an elegant solution. (Jean-Marc: This may also be a good test file to throw into your routine listening) I know I am pushing the codec to its limits a bit: fixed point math, small frame size (100 samples on v0.7.1, 120 samples on v0.11.1) and a bitrate of around 200kbps for a stereo application. I understand I can throw more bits or a larger window at the problem, but then my system requirements would be blown. Would the complexity option help me at all? Has anyone worked around this some other way? thanks in advance -------------- next part -------------- An HTM...

...dn''t understand any of it) #define INTERNET eth0 #define LAN eth1 #define DMZ eth2 #define INTERNET_IP 0.0.0.0/0 #define DMZ_IP 10.2.1.0/16 #define LAN_IP 10.1.1.0/16 #define maxadsl 600kbps/2 #define highadsl 500kbps/2 #define medadsl 400kbps/2 #define midadsl 300kbps/2 #define lowadsl 200kbps/2 #define intadsl 150kbps/2 #define vloadsl 100kbps/2 #define noadsl 50kbps/2 dev INTERNET { $meter = trTCM( cir 128kbps, cbs 10kB, pir 200kbps, pbs 10 kB ); egress { class(<$lanssh>) if tcp_sport == 22 || tcp_dport == 22 if ip_src == LAN_IP || ip_dst == LAN_IP; class(<$dmzssh...

...p://stream.orban.com:80/eval -greg. At 00:21 2004-12-29, qiang Bao wrote: >it seems that icecast 2.2 can only stream aac at 128kbps! > >mp3 and ogg are ok at all bitrates. > >is it a bug or my config wrong? > > I use winamp 5 with oddcast plugin. The plugin showed it sent 200Kbps AAC to icecast,but icecast can only stream out at constant bitrate 128Kbps ¡£ I mean that icecast stream aac at constant bitrate which is 128. 128kbps 32KHz it can stream at 32KHz 44.1KHz 48Khz,i know that. I also want to know how to make it stream at other bitrates? and which source sup...

...with htb where the amount of bandwith could vary.? I have an idea where I can create a paterrn as : Main Link Parent Childs A Child B Child C Child X Where X is the amount of bandwith that could vary? And it change in period of time as from 8:00am - 12:00pm bandwith 200kbps and from 12:00pm-16:00pm 100 kbps. And so on. Depend of the stadistic I assume. And reload the script several time with crob. Does anyone knows a better way? Regards. -- Alex Segura N. Jefe de Proyectos Redes - VoIP <BLOCKED::mailto:a.segura@kernellinux.com> a.segura@kerne...

...may also be a good >> > test file to throw into your routine listening) >> > >> > I know I am pushing the codec to its limits a bit: fixed point math, small >> > frame size (100 samples on v0.7.1, 120 samples on v0.11.1) and a bitrate of >> > around 200kbps for a stereo application. I understand I can throw more bits >> > or a larger window at the problem, but then my system requirements would be >> > blown. Would the complexity option help me at all? Has anyone worked around >> > this some other way?

I'm looking to encode my meager (200+) CD collection as soon as I get a nice, big hard drive to store it all. I was considering using LAME with '-V 1 -h' to get high quality encodes at about 180-200kbps, but obviously I'd like to go for a free(libre/lucre) format. I realize that, in principle, vorbis can encode at ~256kbps, but I'd like to know just when I can expect it to do that level of encoding quality in practice - 3 months? 6 months? Thanks, John --- >8 ---- List archives: ht...

...or ill just put the filter of rtsp protocol to prio 3 instead of 0 .... thanks, does the priority differs from class / ip address / filter etc ? ex: class prio 0 1 2 3 4 thanks, - stephanie here is my partial sample config tc class add dev eth0 parent 1:1 classid 1:10 htb rate 100kbps ceil 200kbps burst 2k prio 1 tc class add dev eth0 parent 1:1 classid 1:20 htb rate 50kbps ceil 100kbps burst 2k prio 2 tc filter add dev eth0 parent 1:0 protocol ip prio 0 u32 match ip protocol 6 0xff match ip dport 554 0xff flowid 1:10 tc filter add dev eth0 parent 1:0 protocol ip prio 0 u32 match ip pro...

I tested vorbis. I sounds good, but when I play something in mono, trebles are much stronger. Perhaps is stereo phase of hi bands removed to improve compression ? I't a good idea (it's not audible in most music), but it should also provide an option to preserve them. Is there docs that explains how dows vorbis compress sound ? oggenc (yearstday's cvs) creates only ~128k streams.

...prio 16 u32 match ip src 192.168.1.39 flowid 1:22 filter add dev eth1 parent 1: protocol ip prio 16 u32 match ip src 192.168.1.40 flowid 1:22 filter add dev eth1 parent 1: protocol ip prio 16 u32 match ip src 192.168.1.47 flowid 1:22 class add dev eth1 parent 1:20 classid 1:23 htb rate 10kbps ceil 200kbps burst 6k filter add dev eth1 parent 1: protocol ip prio 16 u32 match ip src 192.168.1.33 flowid 1:23 filter add dev eth1 parent 1: protocol ip prio 16 u32 match ip src 192.168.1.34 flowid 1:23 filter add dev eth1 parent 1: protocol ip prio 16 u32 match ip src 192.168.1.37 flowid 1:23

...T 98% transparent to the original. Sometimes the piece will be encoded at 200 kbps, sometimes 105 kbps (these are real cases). It'll almost never (98%) contain artefacts, even the latter-105kbps file. Now, the same difficult piece which was classified by Lame's sophisticated algorithms as a 200kbps put to OGG with ABR 128kbps. What the output will be like?: - it'll be encoded at about 145 kbps - not 128 kbps (which is still not enough, even for OGG) - it'll sound as a piece of s... - completely distorted and muffled Of course You may say: "You could have taken 192 kbps in OGG!&qu...

I have a Cable ''modem'' that has a problem that many of these devices is bound to have. I was wondering what other lartc user thought about this and if DSL has simular problems. Cable, being Asymetrical, is able to upload at a given rate and hopefully has a buffer, in the modem, of some kind. When I nc(netcat) to a UDP echo, or discard, server I get about 10Mbps out on my

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Hi. We just opened a gaming center and have chosen to run a FreeBsd box for our firewall. IPFW is configured at it's very basic running natd through rl0 and allowing any to any connections from the lan to the outer world. Natd controls access to the lan. We have a 6.0 mb/s ADSL net connection for all the gaming clients to use, however if a gamer starts downloading a file, that file