search for: 56kbps

Dear All, Sorry to trouble again..... After go through www.lartc.org I have implemented the HTB instead 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.ro...

...ut into mine to open up their little mini-window, but when I do a search for the station it doesn't show up; also if I add it to my presets on their site and then open the presets, it always shows the station as unavailable.. It turns out that if you're streaming at any rate higher than 56kbps, they don't list you in their directory. The "official" word from support is that they don't support rates over that. I was streaming at 96kbps and although as I said, I could listen just fine if I went through my link, nobody searching for me or getting listings from live36...

Hello, Does anybody know how I can configure icecast so it doesn't disconnect a client which "has fallen too far behind". We are using icecast2.01 streaming software for our church home-listening system. Our client we use a netgem-netbox. We are streaming 25kbps MP3 on V90-56kbps modem connection After around 45 minutes after starting the cleint the icecast error.log reports: [2004-11-27 20:25:58] DBUG format-mp3/format_mp3_write_buf_to_client Client had recoverable error -1. But the client still goes on playing for about 1 hour. Then it suddenly stops the music. The ic...

Hi there. I have a short .ogg with Vorbis stream @56kbps from a 44.1 16bits stereo sample, which weights exactly 135'781 bytes. Packed to ZIP, it goes down to 109'485 bytes, that's a 80% ratio ! Even worse with latest RAR3 archive, with which the file size goes to 104'349 (76%). <sceptical> I admit this sample is a bit repetitive,...

...on eth0 I''ve the dsl connection (1.2 Mbit / 256 kbit) I need to limit the bandwith towards lan and I''ve thought at HTB and tcng. I write the script belove. I want limit the p2p and ftp (ssh and irc) connection at 15 kBps (128 kbps) and the other type of traffic at 32 kBps (256kbps). Is the script right? Moreover I want understand better how htb work: I can shape the traffic router->lan but I can''t shape traffic lan->router, is it right? In this way I can limit the bandwidht towards lan, but I have the bandwith towards router always full. Thanks P.S.: sor...

...Ogg didn't measure up as our best candidate. We still think Ogg is a worthwhile codec, and as such SHOUTcast-streamable support for it is available via NSV. While I can't yet announce the precise codec we'll be using for the next 3 year run, I will say it is making me very happy to see 56kbps streams outperforming (from a purely subjective "quality" analysis) current 128kbps streams, and the cost of the encoding seems manageable enough for AOLTW to continue absorbing encoding and decoding costs, not to mention the majority of playback devices and applications will be supportin...

...Ogg didn't measure up as our best candidate. We still think Ogg is a worthwhile codec, and as such SHOUTcast-streamable support for it is available via NSV. While I can't yet announce the precise codec we'll be using for the next 3 year run, I will say it is making me very happy to see 56kbps streams outperforming (from a purely subjective "quality" analysis) current 128kbps streams, and the cost of the encoding seems manageable enough for AOLTW to continue absorbing encoding and decoding costs, not to mention the majority of playback devices and applications will be supportin...

...Ogg didn't measure up as our best candidate. We still think Ogg is a worthwhile codec, and as such SHOUTcast-streamable support for it is available via NSV. While I can't yet announce the precise codec we'll be using for the next 3 year run, I will say it is making me very happy to see 56kbps streams outperforming (from a purely subjective "quality" analysis) current 128kbps streams, and the cost of the encoding seems manageable enough for AOLTW to continue absorbing encoding and decoding costs, not to mention the majority of playback devices and applications will be supportin...

...their little mini-window, but when I do a A> search for the station it doesn't show up; also if I add it to my presets A> on their site and then open the presets, it always shows the station as A> unavailable.. A> It turns out that if you're streaming at any rate higher than 56kbps, they A> don't list you in their directory. The "official" word from support is A> that they don't support rates over that. I was streaming at 96kbps and A> although as I said, I could listen just fine if I went through my link, A> nobody searching for me or gett...

Please pardon me if you have already seen this posting, but I received no postings from the list for a week....I assumed the list was down. I would like to batch re-encode MP3 files on the Icecast host. A command-line utility to take say a 56kbps MP3 file as input and produce a 20kbps MP3 file as out would be ideal. --------------------------------------------------------------- Todd Merriman - Rainbow Radio Sound & Light +1 770 889 5897 8030 Pooles Mill Dr., Ball Ground, GA 30107 dj@RainbowRadio.com http://www.RainbowRadio.com -------...

My needs are simple. I need a Windows client to accept the line input on my sound board and stream it to an Icecast server. I plan to encode the stream at 56kbps stereo. I tried this about 2 years ago using Winamp with the Oddsock plugin. This turned out to be a very time-wasting excersize in futility. I gave up, but now I'm back hoping the technology (and documentation) is there. My server is already running 4 streams, two 56kbps and two 24kbps. They...

...ient which "has fallen too far behind". there isn't anything else icecast can do when that condition is reached > We are using icecast2.01 streaming software for our church home-listening > system. > Our client we use a netgem-netbox. > We are streaming 25kbps MP3 on V90-56kbps modem connection a 56kbps modem is not going to have a lot of tolerance. > After around 45 minutes after starting the cleint the icecast error.log > reports: > [2004-11-27 20:25:58] DBUG format-mp3/format_mp3_write_buf_to_client Client > had recoverable error -1. > > But the c...

...header of 20 bytes and a TCP header of 20 bytes, assuming no options, that leaves 1,460 bytes for payload. 7,000 / 1,460 = 4.8 packest/sec. 4.8 pkts/sec * 40 bytes of header = 192 bytes/sec. 7,192 bytes / sec = 57,536 bits /sec. All-in-all, the IP overhead accounts for an additional 3% with a 56kbps stream. Take the overhead and multiply it by the number of simultanenous clients. In my experience, though the actual data rate is higher. I haven't done any further testing in awhile and I assume that the theoretical maximum of stuff put in each packet isn't the full 1460 bytes. -I

...PSTN side interfaces - atm0 and dialer0 (dialer0 is where the ISP details are configured and the interface that the WAN side IP is applied on) Now which interface to I apply the QOS on? I would have thought dialer0, but for some reason this interface does accept is citing that it only has "56kbps" of bandwidth for some reason. Now I could fix it's bandwidth at 1000kbps, but as anyone who uses ADSL will know, this may not be a good idea as ADSL is rate adaptive, meaning that the bandwidth can increase and decrease due to line conditions. What do I do? I tried "bandwidth in...

...ample rate from speech at 48kHz, much less hear a large quality difference? A number of audiobooks I've listened to have used 22kHz mp3s without being clearly objectionable, and in my personal use I've had decent results using the -voice LAME setting (downsamples to 32kHz and encodes as 56kbps abr). The recent hydrogenaudio tests showed Opus CELT modes trumping the best of breed high-latency codecs at 64kbps despite having only 22.5 ms latency, and the SILK modes do a great job at the opposite of the bitrate spectrum and can make use of larger frame sizes for those of use who don...

At 04:38 PM 12/12/2002 -0500, you wrote: <p>>or , anyone have luck w/ streamTranscoder ? looks promising , but it >won't connect to the output stream's server correctly . I've had very good luck with streamTranscoder...and not just because I wrote it....oh wait...maybe it IS because I wrote it....but seriously, what the problem ? feel free to send me info off list if

...cation, but normally I > would expect an IMAP client to request headers-only, or just recent > messages. My phone's K9 client defaults to something like 2 weeks. Right. But your phone has 100s of kbps or even Mbps of bandwidth, full duplex. But when multiple users are sharing a simplex 56kbps radio channel, even just the headers of old/irrelevant mail are to be avoided. Michael

...hewing 32secs of the slack probably woudn't help either. Background info =============== I'm running an icecast-2.1 mp3 streaming server where a few hundred listeners connect to a few tens of streams at 16 and 24 kbit/s bitrate. Listeners connect using a device which is dialing in using a 56kbps analog modem. Some listeners experience poor stream stability at times, meaning that the stream repeatedly skips (with pauses of around 20-30 seconds). I've looked at various reasons for this behaviour such as bad connect speed of the 56K modem and possible lack of bandwidth on the streaming s...

...ote: > While dynamic compression makes the transmission 'smaller' on FM radio, > it makes music harder to compress by psychoacoustic codecs, and therefor > you might have somewhat of a quality hit, although I haven't tested > this. Quite possible, though I'm only using 56kbps 22khz stereo anyway so that's not such a big deal. I really need it to even out levels, however, which is more important to me. Geoff. -- Geoff Shang <gshang@uq.net.au> ICQ number 43634701 Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-w...

...server (with XMMS or MPlayer) the buffer runs empty very quickly, causing interruptions. My "server" is an old Pentium 120 with 64MB RAM ,a HD of 8G and an ethernet NIC. (I know it isn't much, but I'm tired of taking my CD collection to work.) I ripped the tracks in OGG format at 56kbps 22050 Hz Samplerate. my partial ices config: <encode> <nominal-bitrate>56000</nominal-bitrate> <!-- bps. e.g. 64000 for 64 kbps --> <!-- samplerate>22050</samplerate --> <channels>2</channels>...