similar to: Akos...Darkice questions

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

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

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.

Hi everybody I'm trying to encode the soundcard signal to serve it with IceCast, but I'm having problems with making darkice-0.11. Anybody could help? I'm compiling in a Solaris 6 OS with gcc 2.8.1, and I've installed lame and Vorbis libraries. This is my config phrase: ./configure --prefix=/aplicacions/icecast-1.3.12/darkice --with-lame --with-vorbis

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... >

Kristjan Gu?ni Bjarnason wrote: > I was looking more closely on the source for Darkice. I found the code for > calulating the buffersize somewhat strange. Your is snapshot taken from the > init function in CastSink.cpp : > > int bufferSize = bitRate ? (bitRate * 1024 / 8) * bufferDuration > : (128 * 1024 / 8) * bufferDuration; >

I was looking more closely on the source for Darkice. I found the code for calulating the buffersize somewhat strange. Your is snapshot taken from the init function in CastSink.cpp : int bufferSize = bitRate ? (bitRate * 1024 / 8) * bufferDuration : (128 * 1024 / 8) * bufferDuration; bufferedSink = socket ? new BufferedSink( socket,

I'm compiling Darkice on a FreeBSD 4.6.2 box. Everything works up to this point after I type "make"...then I get this error. Any ideas? Thanks, Matt teststream# make make all-recursive Making all in src source='TcpSocket.cpp' object='TcpSocket.o' libtool=no depfile='.deps/TcpSocket.Po' tmpdepfile='.deps/TcpSocket.TPo' depmode=gcc /bin/sh

Hi, I am trying to set up a stream from my computer. I have set up Icecast, and it seems to be working. My OS is Redhat Enterprise Linux 3 and my soundcard is a SB Audigy SE working with Alsa. I have set up darkice as source client, and it seems to be working I get a really slow input stream from darkice. I think the stream must have double speed for a correct playback. This is the

>Fromthe config.log file created by ./configure configure:3326: checking for sys/types.h configure:3339: gcc -c -g -O2 conftest.c >&5 configure:3342: $? = 0 configure:3345: test -s conftest.o configure:3348: $? = 0 configure:3358: result: yes configure:3627: checking sys/socket.h usability configure:3636: gcc -c -g -O2 conftest.c >&5 configure:3639: $? = 0 configure:3642: test

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.

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

This is probably sacrilegious to speak of Shoutcast here but that's what I am streaming to right now. Just a couple questions about Darkice... quality is set to 0.5 by default. What's the range for this and what kind of effect does it have on the audio and the load on the system? I've set the "public" variable to yes so it will advertise to Shoutcasts directory but

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

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;

Unfortunately I can't reproduce your memory leak reports. For example, here is the line about my production darkice instance from top: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME COMMAND 27516 root 20 0 3564 3564 1340 S 95.6 1.3 47:06 darkice <p>this as after approx 50 minutes from start, but the memory load doesn't change for the whole 4 hours it

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'm running Icecast 1.3.11, Lame 3.89, and the latest version of liveice. I'll post my config files in case someone might find an error in them... ################################## ##### ICECAST.CONF location Radio rp_email matt@cmitech.com server_url http://216.89.xxx.xxx/ max_clients 10 max_clients_per_source 10 max_sources 1 max_admins 5 throttle 10.0 use_meta_data 0

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();

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,