similar to: Writing a client

As you know, Winamp relies on the file extension to know what input plugin to utilize for the audio decoding. The input plugin for .ogg files (in_vorbis.dll) needs to understand the other ogg container formats but it doesn't as it is a vorbis only decoder. So the only solution is for Ogg Speex to have a different extension so this can be appended to the mountpoint, and an input plugin

Karl Heyes wrote: > The handling of metadata is specific to the format, eg MP3 itself has no > in-stream metadata support, only the shoutcast-style streams have > metadata handling. Vorbis has in-stream metadata support but is handled > differently to MP3. I see. so we have separate solutions for: - shoutcast (mp3) - icecast 1.x (mp3) - icecast 2.x - mp3 - ogg vorbis > The

I would like to program a basic OGG decoder that takes in an array of bytes (unsigned char *) and return an array of floats. Presumably I could then pass these floats to be read as PCM data which can be immediately interpreted as audio (by Unity3D). The starting point would be the header of an icecast stream (I know I'm not in the icecast mailing-list, but that is only the basic starting

Hi all, I set up ezstream to read a playlist.m3u file containing only MP3 encoded songs (to shuffle continuously) and push this to icecast2. This works, and I can also reencode to a lower bitrate. I would like to know if there is a way to reencode these MP3s into AAC-LC 128kpbs and then push the stream to icecast2. I'm doing this on a VM with only terminal access and no audio hardware, so

Hi all, I am recently using icecast from server streaming, and even though I succeeding setting everything up to have a radio broadcast, I still have a few questions regarding icecast use. First, latency. I have a setup on a LAN, where I can add audio files to the stream, with a python script that automatically send the location of the file to be played continuously. I wanted to measure the

> You must have some other, unrelated, problem, presumably elsewhere in your > code. this is what i have, and does work with mp3, but ogg does not want to work. im testing with winamp, you think that may be the issue? <p><p>fwrite($fp,"GET /mymp3 HTTP/1.1\nUser-Agent:$HTTP_USER_AGENT\n\n"); for ($i=0; $i<120; $i++) { if (feof($sp)) break; $str.=fread($sp,4096);

oddcast currently doesn't do speex, or ogg-speex. One of the biggest problems currently with speex (and flac too) is client support. I've just recently added ogg-flac support to oddcast (not yet available) and in my testing found that client support for this format is horrible. On windows, Winamp and Foobar choke on it (most likely assuming that if it's ogg, it must be vorbis)

Well the code itself is what i took from tom peppers sc proxy script and incorperated our own auth system for icecast2 i know i got it close with ogg and it seems to be when i tweak the $str.=fread($sp,4096); bit amount, i get different results, 4096 works fine for mp3 on icecast2 and shoutcast, but im assuimg maybe there is more or less to read from the ogg headers?? if so what would be the

this is an ICE protocol question - how would a source client send an update of the stream metadata to an icecast 2 server? as much as I know, the source sends an HTTP-style header at the beginning, with various metadata fields, than an empty line, then starts to send the binary audio data to icecast. but if the metadata for the stream needs to be updated, I guess the source has to 'stop'

Hi, My friend who is not a techie have a small radio and they used to broadcast live with a mic plugged no a Ubuntu PC. On Ubuntu 10.04 they used darksnow with the right config file and it was OK. But now under 12.04 darksnow crash, butt is really difficult to start wit etc.... So simple question, is there any straight forward icecast client for ubuntu 12.04 with a decent GUI ? they

At 05:49 PM 12/14/2002 +1000, you wrote: >Does vorbis input work yet? I'm really keen to know. We're currently >using streamTranscoder to provide modem rate versions of a 56kbps stream, >but I'd really like us to source with a 64kbps ogg stream and transcode >that instead. The news section on the website says that it doesn't work >(or didn't on October 16)

hi ! i have succesfully compiled libshout/icecast/ices on a quad ss20 (4xhypersparc@125mhz, 128mb ram). ten out of ten for a robust, cross-platform build, guys ! it compiled straight out of the box. i can start the server and it even connects to a remote server and starts to relay. when i run xmms from another machine and connect, it stops at "pre-buffering 0/32k". here's an

Hi: Since I'm not a programmer and just want to get stuff worked out properly in my head, I'm posting this here. I just want to be sure I have all the concepts regarding vorbis and high sampling rates at low bit rates all sorted out. Note that this will involve questions, some of which I've been looking for answers to for some time. The way I understand things is that vorbis is

On Friday 17 January 2003 20:17, Karl Heyes shaped the electrons to say: > I would suggest a slightly different approach. > > Instead of increasing the syscall overhead for all sockets, trapping > for uncommon cases. Try the sock_write_bytes and if that is > continuously having to queue (ie not all data can be sent) then display > the warning, maybe make it a run-time option

I'm having trouble with the "Move clients" command in the admin interface. All the clients using JOrbis (a java vorbis decoder) just stop playing, and don't switch to the new stream. I'll assume this is a JOrbis problem rather than Icecast2; has anyone else had trouble with this combination before? Thanks. <p> -- Luke Stodola minus273point16c@fastmail.fm --

There is a big, big difference between cbr and vbr, btw. The machine in question is a Pentium 3 700 oc'ed to 800, 512MB ram, win nt 5.1 (2k and xp being nt) etc In an earlier version the oddcast dsp ate up so much cpu time when set at say 64kbps (for examples sake) 44.1kHz, stereo that the machine became totally unusable and the winamp task had to be (slowly) ended with task manager. Now,

I am totally new to icecast, and would appreciate a pointer in the right direction. My application is to encode audio (at the linux server) at a low bit rate, say ~16 kbps, for transmission to a decoder at another location. The audio source will be bandlimited (basically voice grade circuit such as you'd find in a telephone application.) The decoder end (connected via a TCP/IP session)

Thanks Keith for the pointers on icecast and ices-cc installation how-to. I followed the instructions and completed the two installations in no time. Once, the two programs are installed on CentOS, what's next. Lend a guiding hand to an eager newbie. And keep those straight guidelines coming, please. TIA On Sun, February 13, 2011 17:14, Keith Roberts wrote: > On Sun, 13 Feb 2011, Huw -

Hi guys, I am working on an application that gathers and stores toll-quality/narrow-band voice data. It will allow clients to request this data and stream it to them on the fly. I'm planning on this data all being stored in the speex format (possibly encapsulated in an Ogg file header). I was wondering what method the members of this list would recommend for streaming the data to

Here I leave some source client android for Icecast. There is one that works with mp3, one that works with OGG Vorbis and one more that works with Theora, I've only tried the MP3 and Ogg Vorbis, and have certain limitations but work well. Greetings. Aqu? dejo algunos clientes source android para Icecast. Hay uno que funciona con mp3, otro que funciona con OGG Vorbis y uno mas que funciona