similar to: Remote Telecasts?

hello On Fri, 14 Dec 2001, Leo Currie wrote: > > Does anyone have experience doing remote live broadcasts over Icecast? My > > thought is to use a Dell laptop running Windows (yeah, I know ;-), > digitize > > locally to 16khz, and pump the output to a remote Linux box. > > > > Has anyone done something like this before? Thoughts? Issues? [...] > > Oh -

> Does anyone have experience doing remote live broadcasts over Icecast? My > thought is to use a Dell laptop running Windows (yeah, I know ;-), digitize > locally to 16khz, and pump the output to a remote Linux box. > > Has anyone done something like this before? Thoughts? Issues? We (Student radio station in Glasgow, UK) took a feed from a DJ in Cambridge and re-broadcast it

> > > > the bitrate on the mp3's is 128 (great for listening off the HD) > > > > whats a good bitrate to encode for streaming audio? > > I think this might be the problem 128kbps isn't going to work over a modem. > > I think 24kbps is probably the pratical limit for 56k modems. 56k modems might > be able to get a 32kbps stream but only under

OK. So I am running a 1.5 Ghz P4 with 256mb runing redhat 7.3 using Liveice, Lame and Icecast. Liveice and lame on the source box and icecast on the streamer box. 96k and 56k streams sound wonderful. But the 24k stream sounds like gerbils talking to each other. Any ideas of what would be causing this? Thank You, Mike <p> -- ///////////////////////////////////////// - Mike

On Fri, 6 Apr 2001, John Griffiths wrote: > ok so 24kbps for 56k modems... > > can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of

use icecast+liveice and i need to get stream from linein . Icecast work perfectly and liveice was perfectly but i can get stream only if I plug lineout from my radio box to mic on my sound card . Is there any special config string in liveice for streaming from linein or maybe liveice can't do it my liveice.cfg # liveice configuration file SERVER localhost PORT 80 NAME My Radio Box GENRE

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Greeting all -- I'm having some problems setting up multiple streams with iceS, for example, a hi-bandwidth and lo-bandwidth stream of the same audio. Could someone post a config that works in this case? thanks .oOo.oOo.o..o.oOo.oOo. Ben Wilson admin -- thelocust.org ben@thelocust.org 'OoO'OoO'O''O'OoO'OoO' --- >8 ---- List archives:

On Mon, 2003-08-25 at 17:04, W. Kevin Pedigo wrote: > But if your problem is serving more bandwidth than you've got, you gotta > serve less (narrower or fewer streams) or get more bandwidth. It's that > simple. Tell us what you want to do about it, and we'll try to help. OK. I've gotten everything running with one problem. I'd like to downsample a live stream.

http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKX&version= 0 can anyone suggest how I could set up asterisk@home to read out allowed the following text when I dial extension 850? 815 PM EST WED FEB 16 2005 .OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO 20 MPH WITH GUSTS UP TO 30 MPH...DIMINISHING TO 10 TO 15 MPH LATE. .THURSDAY...PARTLY CLOUDY. COOLER

hi folks, after some adventures for setup an icecast server , the system is now running, but there where some questions open: 1. ) i use ices as streamer for the server. in some cases (i dont know why) is ices break after an while of streaming. ices re-encode the mp3 files with lame 3.88 to stream them. there is no hint why the stream is broken in the log files. after an restart of ices all looks

On Sat, Nov 17, 2001 at 06:57:46PM +0100, Maroy Akos wrote: > On Fri, 16 Nov 2001, Samuel Hathaway wrote: > > > I agree! Also, something I've been looking for is a way to pull sound from > > the dsp device at 44kHz and then downsample it to 22kHz for one of my two > > streams. Ideas? > > DarkIce does this already. What it doesn't do, is to have mono and

Hi, Sorry to be posting about a subject that may have already been answered. If so, please point me in the right direction. I'm developing a dictation application on the Dell Axim (Windows Mobile 5.0 Pocket PC). A key requirement of the application is the best possible sampling rate as the audio goes into a speech reco system. So, I've set up my wrapper around libspeex to capture audio

Like any other major radio station we output our music from automation software via analog audio through a mix board in a studio, where we insert other stuff like live DeeJays, etc. only to have that stream re-encoded by hardware MP3 encoders for distribution to our network of IceCast servers. The chain of song information gets broken as soon as we output from the Automation software,

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

Ok, is this possible: I want to have a 128k and a 24k stream of a particualr audio program, plugged into the line in of my Ensoniq AudioPCI 128 (es1370 chipset) Is there any way to do this with just one soundcard, or do I need two? Thanks Scott W --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a

Hi friends, I haven't done such a simulation before and any help would be greatly appreciated. I need your guidance. I need to simulate end to end data for Reliability/survival analysis of a Pump ,with correlation in place, that is at 'Transactional level' or at the granularity of time-minutes, where each observation is a reading captured via Pump's sensors each minute. Once

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure