similar to: darkice slow stream

We're running a darkice/icecast configuration to stream our LPFM radio station (and to provide a quick-n-dirty studio-transmitter-link). (See config data below.) We're getting intermittent "Buffer overrun!" errors out of darkice, and we associate these with interruptions in our audio stream that are quite noticeable. These occur anywhere from once to ten times per hour.

hello, I need to set up icecast2 as indipendent server, I mean I have my own domain and IP address where I'm going to install icecast2, my server have to be able to authenticate every SourcesSide (eg: winampa, darkice) but can't udersand something, as like as difference between "master server" and "relay server" and, in my darkice confinguration file, where i have to

Ok, I'm using icecast2 (latest cvs as of maybe 3 hours ago) and darkice .9.1 all compiled under debian/sid with gcc 2.95.4. Everything works ok, as long as I'm using ogg123 to listen to my stream. using xmms (1.27 in in debian sid) it'll buffer up to 50% and then start over again seemingly indefinitely. with winamp 2.80 in winxp, I get nothing but silence, I'd normally thing

I recently installed icecast and iceS and all seems to be well. I have spent four days trying to understand and use darkice and finally gave up. Using darkice configuration: [general] duration = 0 # do it forever bufferSecs = 10 [input] #device=hw:0,0 device = /dev/dsp sampleRate = 22050 # 44100 or 22050 or 11025... bitsPerSample = 16 #16 bit samples channel = 2 # 1 for mono, 2 means stereo.

Hello, I'm useing freebsd 5.2.1 with icecast 2.0 and darkice 0.8 with a radiocard from avermedia 206 and a cmedia soundcard. i't works fine! just four points. 1. the ogg with vbr and a quality of 0.6 has a high tone in it. is there a known reason for? 2. filerecording works fine but how can i make every hour a new file without breaking the stream for listener? i thought

Thanks, did anyone made a kind of GUI for ices ? <-----------------------------------------------------------------------------------------------------------> web perso : http://memeteau.org Boutique Ordinateurs GNU/Linux : http://shop.ekimia.fr <xmpp%3Afreechelmi at jabber.fr> 2012/11/14 Jos? Luis Artuch <artuch at speedy.com.ar> > ** > Michel, with

Can the bitsPerSample be set any higher than 16? I try to set it to 20 or 24 and get this error... DarkIce: LameLibEncoder.h:122: specified bits per sample not supported [24] teststream# <p>When I start Darkice I get the following error for about 5 seconds and then it stops with "broken pipe". 08:57:09: BufferedSink, new peak: 53112 08:57:09: BufferedSink, remaining: 28808

Actually...I think it's because I set it to mono instead of stereo...anyway, is there a way to set the bitsPerSample higher? Matt <p><p>>>> Matt@cmitech.com 8/28/02 9:09:03 AM >>> Hmmm...I raised my sample rate to 44100 from 22050 and now it seems to be more stable??? >>> Matt@cmitech.com 8/28/02 9:04:47 AM >>> Can the bitsPerSample be set any

On Tue, 2004-06-08 at 12:12, kevin wrote: > I am trying to relay a streamtuner Live365 stream across a ssh tunnel to > a remote XP pc, where I would pick it up using winamp. > > I have a similar setup with a non-live365 stream routed using > streamripper to port 9001 and this plays correctly in winamp > > For my Live365 stream, I have a RealVNC root session that is streaming

I am trying to relay a streamtuner Live365 stream across a ssh tunnel to a remote XP pc, where I would pick it up using winamp. I have a similar setup with a non-live365 stream routed using streamripper to port 9001 and this plays correctly in winamp For my Live365 stream, I have a RealVNC root session that is streaming the Live365 broadcast from streamtuner (using XMMS). I can see XMMS playing

? am newbie about icecast and darkice. Here is my informations about runnings and configs. Can u help me what i have to do in configs? -su-2.05b$ icecast -b -c /usr/local/etc/icecast.xml Starting icecast2 Detaching from the console -su-2.05b$ [2008-01-19 10:27:19] INFO main/main Icecast 2.3.1 server started [2008-01-19 10:27:19] DBUG yp/yp_recheck_config Updating YP configuration [2008-01-19

Hmmm...I raised my sample rate to 44100 from 22050 and now it seems to be more stable??? >>> Matt@cmitech.com 8/28/02 9:04:47 AM >>> Can the bitsPerSample be set any higher than 16? I try to set it to 20 or 24 and get this error... DarkIce: LameLibEncoder.h:122: specified bits per sample not supported [24] teststream# <p>When I start Darkice I get the following error

At present my stream is at 11.025 kHz and I want it to be at 44.1 kHz. Input is coming from line-in on my sound blaster card under linux (RH 9) using the sb driver. I presume that it is icecast that sets the sample rate on the dsp in the card, though when I change the settings in icecast.xml and darkice.cfg as show below the stream becomes choppy; or rather the sampling doesn't seem to

I'm using liveice and lame on another stream... ample rate of 22050 bitrate of 24000 It has worked fine for over 9 months now...except it crashes all the time because of liveice I think. >>> darkeye@tyrell.hu 8/28/02 9:15:58 AM >>> Matthew Mencel wrote: > Can the bitsPerSample be set any higher than 16? I try to set it to > 20 or 24 and get this error... >

Earl Chew wrote: > I'm a little reluctant to introduce another compiled program when there are > so many other options that will work well enough out of the box. > > Here are two ideas: > > 1. Use bc(1) to compute the raw samples > 2. Use perl(1) to compute the raw samples > > To generate raw unsigned samples using bc(1) for example: > > samplerate = 1000;

I'm using seekable_stream_decoder, And., this is my write_callback. I'm not getting the required output. The PCM i get is not the proper music. Am I doing something wrong here? FLAC__StreamDecoderWriteStatus AFLACStreamPlayer::StreamWriteCb ( const FLAC__SeekableStreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data) { int Channels,

Erik, It turns out bc(1) is too accurate, and a little slow, for this purpose. I've switched to using awk(1) which uses floating point. Do you feel I need to test for the presence of awk(1) ? It is specified as one of the standard commands in the LSB : http://refspecs.linuxfoundation.org/LSB_1.0.0/gLSB/command.html Earl ??? awk -- ' ??? BEGIN { ??????????? samplerate = 8000;

Here's how I set up the data for processing: // For moving data into 32 bit shape uint8_t *buffer8 = NULL; uint16_t *buffer16 = NULL; uint32_t *buffer32 = NULL; unsigned sample32; unsigned sample, channel; uint32_t bitsPerSample = this->get_bits_per_sample(); numFrames = inData.GetSize();

On Nov 26 22:39:27, hans at stare.cz wrote: > ./test_replaygain.sh fails for me in tonegenerator(), saying: > > ./test_replaygain.sh[91]: mawk: not found > Testing FLAC replaygain 8000 (8000 x 1) ... -: ERROR: got partial sample > > Apparently, the tone-generating awk script does not work with > my system's awk, which is "awk version 20110810" as distributed

Brian Willoughby wrote: > What about using the C library sin() and cos() functions to generate > the test audio instead of sox? I did not see a description of how > the test files are generated, so maybe this is easy or maybe it is > hard. The benefit of shipping the test audio generation source code > around with the FLAC sources is that the tests won't break when