similar to: Bit pealing and streaming

Hi all: I am managing a radio station that has 7 streams. I am using icecast, ices and ezstream on the streams and overall it is working pretty well. One of the streams is an old time radio stream and I use ezstream to run the prerecorded shows. The streams are 128K mp3 streams. Also this station is running on a VPS using Ubuntu 14.04. My question: Is there a package or some sort of

Hi: Actually, if the stream were to peal to the best quality dependent on a listener's bandwidth, the capability would have to be in icecast, surely. Geoff. <p> -- 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 avoid sending me Word or PowerPoint attachments. See

Hi, Maybe a bit off topic for this list, bt anyway. I have received several feature requests for DarkIce to support downsampling of the audio input before passing it to lame or ogg vorbis. For example the audio read from the soundcard would be 44.1kHz, and lame would get it at 22.05kHz. I figure two ways of doing this: 1. For lame, one can specify the input and the desired mp3 sampling rate,

Hi everyone, I thought long and hard (ok, not *that* long) about whether or not to crosspost this to icecast-dev. In the end I decided not to since this isn't about an Icecast or Xiph project. My name is Geoff Shang and I've been an enthusiastic user of Icecast for a long time. I am also the chief tech guy for a project called ACB Radio (http://www.acbradio.org), which is run by

David Baelde wrote: > You didn't describe much of your bugs or the specific features of your > playout system... hmmm, I thought I did describe the playout system, but I'm happy to elaborate if you want more details. But you're right about the bugs. The critical (i.e. urgent) ones in no particular order: 1. PRS will disconnect from Icecast after a certain amount of time

Hi everyone, Ok, so I admit I'm asking this to avoid having to do my own research. but I'm also asking in the hopes of avoiding doing my own scripting. ACB Radio is looking at moving across to Icecast. There's really not a lot to stop us at this point. But we are keen to retain access to total listener hours, average listen time and amount of data served per mountpoint. I

Hi,guys, During a HVM''s running which take a qcow format image file as its own virtual disk, the qcow image file will be always read.In the situation that its qcow format image is based on a raw format image, if nesethe backingfile ,just that raw format image file,would be read .my purpose is to cache the data that is read from the backingfile when the hvm is running . Now what I concern

Hello, I have a question with password never expired flag during changing password. If X flag for password never expired has been set in account flags for a user, password change would cause to lose the X flags. By taking a look at the code of the line 993 in passdb/passdb.c, it said all other acb flags will be inherited from current existing account ctrl bit, except for

Hi, I'm trying to relay a shoutcast-style MP3 stream but am getting the following error: [2008-02-05 23:05:05] EROR format-mp3/mp3_get_filter_meta Incorrect metadata format, ending stream Is there any way I can tell what is wrong with the data we're getting? The stream I'm trying to relay is http://acb.org:8550/ This is a shared server so it's possible that not all

Hello, I am setting up a new (remote) server. I initially installed 9.2 RC4 amd64 because that's what the data center put in the drive for me. Shortly thereafter I downloaded 9.2-STABLE sources and compiled world and a generic kernel. While doing so the system became unreachable by SSH The SSH sessions appeared to connect but there was never any data returned. I could telnet to port 22

On Tue, Sep 06, 2005 at 09:56:06PM -0500, Jay Krivanek wrote: > Yea, don't get me wrong, ogg is definitely an awesome format but aacplus v2 > is currently the leader in high fidelity quality at low bitrates. 24kbps > aacplus even kicks ass in my view. Is that 24kbps AACPLUS a full 44.1kHz stereo signal with frequencies preserved up to at least 15kHz? (Just curious, not

I have run into a problem loading a just saved R object using R-devel. I have been saving and loading this particular type of R object for a long while and never ran into this problem. I save, then immediately reload (to test save) and get "ReadItem: unnknown type 65". This error is reproducible after logout from server and restart of emacs and R. Below is my output and

I have run into a problem loading a just saved R object using R-devel. I have been saving and loading this particular type of R object for a long while and never ran into this problem. I save, then immediately reload (to test save) and get "ReadItem: unnknown type 65". This error is reproducible after logout from server and restart of emacs and R. Below is my output and

I tried to get winbind to populate my passwd file. No luck! Any suggestions much appreciated. I get: wbinfo -t Secret is good wbinfo -u 0xc000006d (was 0xc0000022 before using administrator account) wbinfo -g 0xc000006d (was 0xc0000022 before using administrator account) rpcclient SERVER -U % -c querydispinfo cmd = querydispinfo index: 0x1 RID: 0x7d0 acb: 0x10 Account: andyj Name: Andrew

Hi: Well, up until about a week ago, we had our stream up and running on 3 seperate mountpoints and it was rock solid. then we started hitting problems with disconnects and the like. I recompiled everything this morning and set it running again. But one mount has died already. We have 3 streams running. the first is the ogg files as they were encoded (egoplay.ogg). the second is a

At 04:08 PM 10/9/02 +1000, you wrote: >Hi: > >Well, up until about a week ago, we had our stream up and running on 3 >seperate mountpoints and it was rock solid. then we started hitting >problems with disconnects and the like. I recompiled everything this >morning and set it running again. But one mount has died already. > >We have 3 streams running. the first is the

Dave wrote: > 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. This is expected.

Happy New Year from the Xiphophorus Team. It took longer than we thought, but I think everyone will agree this is our best release yet. With drastic quality improvements and new bitrate management features, this release brings us one step closer to 1.0. Along with all the lovely VBR modes you are used to, you now have millions more. oggenc's quality settings are 0-10 in increments of

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

https://bugzilla.netfilter.org/show_bug.cgi?id=1154 Bug ID: 1154 Summary: Allow include statement to operate on directories and/or wildcards Product: nftables Version: unspecified Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: nft