IceStreamer

A simple live audio streamer, intended as a replacement for DarkIce.

IceStreamer supports only a single input source, which it can encode to multiple formats, in parallel, and send to multiple servers. It aims to cover the simple usecase of streaming audio for a web radio station.

Robustness in IceStreamer is of major concern, so the most important feature perhaps is that it never stops when there are network errors while sending. If a certain stream is failing to send, it will reset its state and try establishing a new connection after a while. Note, though, that this feature does not work very well with GStreamer versions prior to 1.13.1, as it takes way too long to timeout (see https://bugzilla.gnome.org/show_bug.cgi?id=571722 for details).

Supported capture interfaces:

Jack
ALSA
PulseAudio
PipeWire

Supported encoders:

Ogg/Vorbis
Ogg/Opus
Webm/Vorbis
Webm/Opus
MP3

Supported streaming servers:

IceCast (1.3.x and 2.x)
ShoutCast

Dependencies

IceStreamer is built using GStreamer. For compilation, you will need the core GStreamer headers installed and at runtime you are also expected to have the following plugins installed:

From gstreamer-plugins-base:

vorbisenc
opusenc
oggmux
alsasrc
audioconvert
audioresample
audiotestsrc

From gstreamer-plugins-good:

shout2send
webmmux
pulsesrc
jackaudiosrc

From gstreamer-plugins-ugly:

lamemp3enc

Configuration

By default IceStreamer reads configuration from a file called icestreamer.conf in the current working directory. Alternatively, you may specifiy a different configuration file with the -c/--config command line switch.

The configuration file should be in the following format:

[input]
# Supported sources: jack, alsa, pulse, test, auto (the default)
source=alsa

# Here you can set any properties of the source element:
# jackaudiosrc, alsasrc, pulsesrc, pipewiresrc
# In this example, device is a property of alsasrc.
# See 'gst-inspect-1.0 alsasrc' for documentation
device=hw:0,0

# Additionally, input format, channels & rate can optionally be set here
#format=S16LE
#channels=2
#rate=48000

[stream1]
# Supported encoders: opus, vorbis, mp3
encoder=opus

# Supported containers: ogg, webm
# Note that this has no effect when encoder=mp3
container=ogg

# Here you can set any properties of GStreamer's shout2send element
# See 'gst-inspect-1.0 shout2send' for documentation
ip=rs.radio.uoc.gr
port=8000
password=<censored>
mount=test.ogg
streamname=Test
description=Test stream
genre=Misc
url=http://rs.radio.uoc.gr:8000/test.ogg
public=true

# You can also include properties of vorbisenc, opusenc, lamemp3enc, oggmux, webmmux
# In this example, bitrate is a property of opusenc, expressed in bps.
# See 'gst-inspect-1.0 opusenc' for documentation
bitrate=128000

[stream2]
encoder=mp3

ip=rs.radio.uoc.gr
port=8000
password=<censored>
mount=test.mp3
streamname=Test
description=Test stream
genre=Misc
url=http://rs.radio.uoc.gr:8000/test.mp3
public=true

# Properties of lamemp3enc
# See 'gst-inspect-1.0 lamemp3enc' for documentation
target=bitrate
cbr=true
bitrate=128

Building

This project uses autotools for building. It requires running autoreconf before configuring in order to generate the rest of the autotools build files that are not under revision control.

$ autoreconf -fi
$ ./configure
$ make