search for: 48kbps

...9;m using IP Masquerading to route internet to the LAN eth0 - LAN interface eth1 - Internet interface this is my firs htb script: #!/bin/bash tc qdisc del dev eth1 root handle 1: tc qdisc add dev eth1 root handle 1: htb default 30 tc class add dev eth1 parent 1: classid 1:1 htb rate 25kbps ceil 48kbps #tc class add dev eth0 parent 1:1 classid 1:10 htb rate 4kbps ceil 48kbps tc class add dev eth1 parent 1:1 classid 1:10 htb rate 4kbps ceil 48kbps tc class add dev eth1 parent 1:1 classid 1:11 htb rate 4kbps ceil 48kbps tc class add dev eth1 parent 1:1 classid 1:12 htb rate 4kbps ceil 48kbps tc cla...

Ok - here's what I want to do... I compiled and installed icecast 1.3.11 and ices 0.2.2, and can get a single lame-encoded stream working. What I'd like to do is set up two streams - one at 48kbps, one at 128kbps - from one source. Reason being is that I'd like to listen to the high-bandwidth stream locally (ie: in another room of the house) while I listen to the 48kbps stream elsewhere. Theoretically, I could set up a playlist, two separate icecast processes, and two separate ices pro...

I thought this might be interesting to you, it's an extract from the latest streaming media newsletter, I was intrested to note that MS are claiming cd quality audio at 48kbps whichi is obviously very low, I didn't think much of it at first because nobody uses MS codecs for audio anyway! However (a bit I missed off this quote by the look of it) I then read mention of portable audio players and thought hmmm, if you can get cd quality audio in a third of the memory, pe...

...stical data to Real that should stay private. I'm talking about free as in freedom, not as in "free beer". Apart from that, Real's software is a nuisance to use. It DOES perform well technically, e.g. in terms of quality at low bitrates (44kbps sounds almost as good as Vorbis' 48kbps ABR), but it's ressource hungry and not trustworthy, too. I wouldn't install and run any server that I (and everyone else) don't know about what it actually does, for both security and privacy reasons. I don't trust Real just like I don't trust Microsoft. You seem to be happy wi...

...n't appear on yp directories (which are of course commented out in the icecast config file) even with my public IP address or a dyndns domain - the stream lags on the internet. Clients bufferize every ten seconds (with only one listener!) I resample the ogg stream on ices from 128kbps to about 48kbps but it changes nothing! is it due to the client which handles poorly ogg stream? Do I need more bandwith? I have xDSL with about 512K of upload bandwith with in fact 96-128K available for icecast. What is the needed bandwidth for every listener and for a standard quality ? Please try it on http://...

An idea would be like for WMA 9 lower bitrates (32-42-48Kbps) to use a 'crystallizer'; which is basically a harmonics booster focussed at transposing sharp tones some octaves higher. Eg: A file has been recorded @ 20khz computer (or 10khz real life) to preserve space. While playing back the file sounds a bit mushy, almost as if someone was speakin...

Hi: I guess this is mainly directed at Monty. I see from the CVS list that a lot of changes have been going in lately, especially in the area of 44.1 and 48khz. I know I was certainly under the impression that this area was more or less complete. My question is, apart from the experimental 48kbps stereo mode that went in today, are the changes for 44.1 and 48khz going to result in quality changes or is it just simpler code? Geoff. -- Geoff Shang <gshang@uq.net.au> ICQ number 43634701 Make sure your E-mail can be read by everyone! http://www.betips.net/etc/evilmail.html Please av...

Hi all, Please guide me How to change frame_size of opus codec at run-time (20ms, 40ms, 60ms) I'm stucking in this case: 1. init codec width default config (frame_size =20ms, bandwidth=48KHz, bitrate = 48kbps...), then in runtime changing: - bitrate = 24, 16, 6kbps: sound is OK - frame_size = 40ms, 60ms: Not OK, sound is distort so bad 2. init codec with frame_size = 40ms , others is default, then in runtime changing: - bitrate = 6kbps: OK - bitrate = 24, 48kbps: Not OK I'm using opus_encoder...

On a PII233 system running linux with ogg-vorbis-RC2, when playing a 2-min 48kbps ogg file, ogg123 takes much more CPU time than XMMS (several seconds with ogg123 vs. less than one second for all XMMS threads added up). Since the pids are the same, I am sure that XMMS did not create new threads when playing one song. I think this has to do with soundcard interaction. <p&gt...

Caleb wrote: > i dont understand you people. >-q0 should be poor quality, only that in vorbis, the poor quality is > actually good! :) Another advantage I see in reducing the nominal bitrate for q0 to 48kb/s with a ~13khz lowpass is a smoother transition in average bitrate and frequency resolution from 22khz to 44khz sampling. Currently there is a jump from 11khz (at 22khz

...<icecast@xiph.org> Sent: Friday, December 30, 2005 2:23 PM Subject: Re: [Icecast] streaming to dialup users gives low quality audio > This doesn't specifically answer your question, but is it possible for > you to use an ogg stream? > > I've been running a stream using a 48Kbps ogg (i'm sure I could push > it lower too...) and it sounds to me and other people pretty much just > as "nice" as a 128Kbps mp3. We can't notice it being audibly worse. > > -Daniel > On 12/30/05, Dave <dmehler26@woh.rr.com> wrote: >> Hello, >>...

Hello, I am trying to enable call forwarding feature on asterisk 1.6.0.9 with asterisk-gui. Sure there is no menu for that on gui but, when i try to write some example scripts to extensions.conf to make it work. I totally failed. I dont wanna install smthing like freepbx etc on the system so, i need help to add unconditional etc call forwarding feature for 1.6 Thanks

I seem to be having problems with people accessing an icecast stream from one of my computers. I am using the current cvs of icecast2 (icecast module), libshout2 (libshout module), and ices2 (ices module) and the RC2 version of the ogg and vorbis libraries. I connect to the internet via DSL and have the DSL modem attached to a Linksys router. The router is configured to forward all port 8000

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

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.

..., This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps G7221_C.24kbps G7221.24kbps G7221_C.32kbps G7221.32kbps G7221_C.48kbps I can get Asterisk to work with G722 and the sound is superior compared to uLAW. I tried to get it working wi...

Would someone mind giving me some feedback on rendering threaded comments: http://io.urbanape.com/io-podcast-part-2-48kbps.mp3

...fits 33.6 kb/s modem > users... If you downsample to 22.05kHz, that mode will produce much smaller files. It will also get way smaller if you use mono input rather than stereo. Vorbis currently does not do this processing for you. I believe the lowest mode that 1.0 will support at 44.1kHz is 48kbps. In any case, you really should re-ask this question on the Vorbis list, where many more people can offer you input. Once more tunings are in, and the tools do things like downsampling and downmixing for you, you'll probably not notice a big difference from what you are used to, except now an...

...Bit upstream to serve all those listeners at that quality. Reduce it to something near (Upstream-~10%)/Bitrate if you plan to spend all your upstream on icecast. I found 32kbps/mono/44.1kHz to be a great low-bandwidth stream with pretty good quality, btw. Unfortunately, it usually hovers around 42-48kbps which means that I can't serve more than 2 listeners reliably with my 128kbps upstream. ;P Moritz -- _______________________________________________________________________ "They who would give up an essential liberty for temporary security, deserve neither liberty or security&...

...Real that should stay private. I'm talking about free as in freedom, not > as in "free beer". Apart from that, Real's software is a nuisance to > use. It DOES perform well technically, e.g. in terms of quality at low > bitrates (44kbps sounds almost as good as Vorbis' 48kbps ABR), but it's > ressource hungry and not trustworthy, too. I wouldn't install and run > any server that I (and everyone else) don't know about what it actually > does, for both security and privacy reasons. I don't trust Real just > like I don't trust Microsoft. You...