search for: 80kbps

...ad of CBQ for the same scenario. Now following files are under /etc/sysconfig/htb directory. eth0 DEFAULT=30 R2Q=10 eth0-2.root RATE=256kbps BURST=25k eth0-2:10.comp1 RATE=120kbps BURST=12k PRIO=0 LEAF=sfq RULE=192.168.200.0/24 eth0-2:20.comp2 RATE=80kbps BURST=8k PRIO=1 LEAF=sfq RULE=192.168.100.0/24 eth0-2:30.server RATE=56kbps BURST=6k PRIO=3 LEAF=sfq RULE=203.145.134.120/29 -------------------- eth1-2:30.root RATE=56kbps BURST=6k eth1-2:30:300.all RATE=56kbps BURST=6k PRIO=3 LEAF=sfq RULE=203.145.134.120/29 MARK=3...

I compared quicktime aac and vorbis,i think quicktime aac is better than vorbis at 80Kbps. please tell me if i'm wrong. using command: qtaacenc.exe --tvbr 31 --highest --samplerate keep test.wav qt.m4a oggenc2.exe --raw -q 1.6 test.wav -o vorbis.ogg the version: qtaacenc version 20100725 with QuickTime 7.6.7 OggEnc v2.87 (libvorbis 1.3.1) links: qtaacenc:http://tmkk.hp.infoseek.c...

Hello, is somebody working on a good low-sampling rate / low-bitrate mode? I encoded today a mono/16KHz/16bit WAV (a TV-talkshow), using OggDrop. The quality of the '64kbps' mode was unacceptable, so I had to use '80kbps' mode. The bitrate averages around 42 kbps, which I found a bit high for this quality. In your opinion, what bitrate should I expect as Vorbis matures? 24 kbps? Cheers, SyP -- I'm not opinionated, I'm just always right! --- >8 ---- List archives: http://www.xiph.org/archives/ O...

80kbps test stream using oddcast DSP and new test win32 port of icecast2 server.... http://www.djlithium.com:8064/djlithium.ogg oddsock --- >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 'vorb...

My last sentence still aplies to the bitrate. We usually use 80kbps, but sometimes we have used 64kbps (e.g. when sourcing from a cellular network). With music you can get the difference, but with voice there's no need to use more bitrate unless you want more quality in the audio itself. -- Xabier Oneca_,,_ El 27/02/2013 15:19, "Jos? Luis Artuch" &l...

hi, I have strange problems with HTB and several hundred classes flat structure i.e. root |--50kbps |--30kbps |--50kbps |--80kbps |--100kbps .... several hundred classes like this Ceil is the same as rate. The machine get no more than 2-3% average cpu(2.4Ghz pentium). What happens is that from time to time the traffic got "stalled". I tried numerous things to solve the problem - cable-wiring, different cards t...

...ults in silence. The problem mp3s were encoded with lame --alt-preset cbr 40 and lame --alt-preset cbr 80. You can access some example mp3s here: http://www.svaudio.org/test1.mp3 (works) http://www.svaudio.org/test2.mp3 (doesn't work-40kbps) http://www.svaudio.org/test3.mp3 (doesn't work-80kbps) My DNS is flaky sometimes, so use 68.53.92.187 if you have problems. All of the problem mp3s work fine when played independently, and mp3_check finds no problems. I realize that I could just reencode these, but I want to know what settings to avoid and would like to avoid having to do that if a...

Since RC2 was released, I've started re-building the music library I left behind at my previous job (all my stuff from CDs, ripped into MP3s). Recently I started an aggressive campaign of converting my CDs to vorbis files... I'm doing about 6 or 7 CDs a day at work =) Yesterday I was listening to my Talk Talk album, and noticed that on this one live track, the stereo seemed to have

El 27/02/2013 16:12, "Jos? Luis Artuch" <artuch at speedy.com.ar> escribi?: > > We usually use 80kbps, but sometimes we have used 64kbps (e.g. when > > sourcing from a cellular network). > > > > With music you can get the difference, but with voice there's no need > > to use more bitrate unless you want more quality in the audio itself. > > > > We use analog...

I'm rebroadcasting a realplayer stream and there are two problems. The icecast2 sounds tinny and its delayed something like 3secs. Any advice to perhaps use speex as it's talk radio? Can I cut the delay? Here is my ices2 config: http://static.natalian.org/2006-06-21/ices-alsa.xml Here is details of the feed, so you can take a listen:

...boost to mp3 is too much to catch. Of course, in a month or so, we'll have mid/side stereo and then the winner will be clearly Vorbis across the board. I'd simply hoped to win on non-coupled stereo alone. Oh, well. (OTOH, the fact that a 64kbps mono Vorbis stream sounds as good as a 80kbps mono mp3 is cause for happiness. I just wanted more :-) Also, the current modes setup is a temporary hack. For one thing, submitting a mode description in a huge struct hardwired into the lib will present a binary compatability straightjacket. I'll be changing that aspect of the API. La...

hello again, I got some just preliminary results .... hope someone can explain them to me... As I already told I have the following config : egress { class (30kbps) {sfq}; class (50kbps) {sfq}; class (80kbps) {sfq}; class (30kbps) {sfq}; class (50kbps) {sfq}; ...hundreds like this... class (10kbps,default) {sfq}; } What I got is traffic starvation very often for a period of ~30 sec.. as proposed I''ve done : egress { ...hundreds classes... class (10kbps, prio 7, default) {sfq}; } It s...

Hi i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. I'll have some simplty IVR business logic and the some queues. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 Support this? Would you suggest me a particular products? The server and the agents will be in the same LAN,

...affic with lower prio (radios, vcn, x11...) # 1:30 bulk (http, ftp, cvs...) # 1:40 the rest (p2p mostly) tc qdisc add dev $IFOUT root handle 1: htb default 40 tc class add dev $IFOUT parent 1: classid 1:1 htb rate ${CEIL}kbps ceil ${CEIL}kbps tc class add dev $IFOUT parent 1:1 classid 1:10 htb rate 80kbps ceil 80kbps prio 0 tc class add dev $IFOUT parent 1:1 classid 1:20 htb rate 10kbps ceil 100kbps prio 1 tc class add dev $IFOUT parent 1:1 classid 1:30 htb rate 400kbps ceil ${CEIL}kbps prio 2 tc class add dev $IFOUT parent 1:1 classid 1:40 htb rate 10kbps ceil ${CEIL}kbps prio 3 tc qdisc add dev $...

I have looked at the codec's available but I don't know how get the highest quality recorded message. If a user calls in over the normal telephone network is this limited to the carriers codec or the codec at the asterisk side? Would I get a higher quality result using VoIP rather than the normal network? Any help would be appreciated. Thanks Fats.

Hi, One of my customer needs per flow rate control, so I write one. The code I post here is not finished, but it seems to work as expected. The kernel patch is agains kernel 2.6.11, the iproute2 patch is against iproute2-2.6.11-050314. I write the code in a hurry to meet deadline. There are many other things to do ahead for me. The code is written in 2 days (including read other

...ated stream's bitrate (according to XMMS and WinAmp) > varies between 49 and 59. Currently min and max are ignored by the library, since bounded bitrates aren't going in until rc3. Also, if you pick 96kbps, that picks the 96kbps mode (if there is a 96kbps mode. It might be picking the 80kbps mode. You'll have to look at the vorbisml to figure out which mode that's actually using). In any case, unlike mp3, which must be at a fixed bitrate, Vorbis won't use more bits than it needs. So often you'll get streams that are less or more than what you picked. That's the...

El mi?, 27-02-2013 a las 15:36 +0100, Xabier Oneca -- xOneca escribi?: > My last sentence still aplies to the bitrate. > Thanks Xabier, I mixed "burst" with "bitrate", was my confusion ;) > We usually use 80kbps, but sometimes we have used 64kbps (e.g. when > sourcing from a cellular network). > > With music you can get the difference, but with voice there's no need > to use more bitrate unless you want more quality in the audio itself. > We use analog mixers where we enter the differ...

Clicking on the "For more information, click here" link on the Digium site nice brings back up the same page I was looking at before, without any additional G.729 information that I can see. I'm wondering if some kind asterisker out there could provide us neophytes with some "typical scenarios" where that codec would be useful to us. For instance, I assume that it

Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___________________________________________________________________________