[Icecast] Re: high and low bandwidth stream
news4 at earthsong.free-online.co.uk
Sun Aug 29 17:21:11 UTC 2004
On Sat, 07 Aug 2004 22:34:05 -0400, dave wrote:
> I've got icecast2 and ices0 going and i've got a stream of mp3 files. I
> don't think i've got all the bugs worked out though. Firstly, could someone
> either explain or point me to a site that explains bitrate and sample rate?
> I want to offer this stream for both broadband and dialup use. I'm sitting
> on a cable connection, but i don't want icecast to eat up all my bandwidth.
> Here's my stream definition in ices.conf.
> <!-- Hostname or ip of the icecast server you want to connect to -->
> <!-- Port of the same -->
> <!-- Encoder password on the icecast server -->
> <!-- Header protocol to use when communicating with the server.
> Shoutcast servers need "icy", icecast 1.x needs "xaudiocast", and
> icecast 2.x needs "http". -->
> <!-- The name of the mountpoint on the icecast server -->
> <!-- The name of the dumpfile on the server for your stream. DO NOT set
> this unless you know what you're doing.
> <!-- The name of you stream, not the name of the song! -->
> <Name>Local MP3 Stream</Name>
> <!-- Genre of your stream, be it rock or pop or whatever -->
> <Genre> Music</Genre>
> <!-- Longer description of your stream -->
> <Description>Favorite Music</Description>
> <!-- URL to a page describing your stream -->
> <!-- 0 if you don't want the icecast server to publish your stream on
> the yp server, 1 if you do -->
> <!-- Stream bitrate, used to specify bitrate if reencoding, otherwise
> just used for display on YP and on the server. Try to keep it
> accurate -->
> <!-- If this is set to 1, and ices is compiled with liblame support,
> ices will reencode the stream on the fly to the stream bitrate. -->
> <!-- Number of channels to reencode to, 1 for mono or 2 for stereo -->
> <!-- Sampe rate to reencode to in Hz. Leave out for LAME's best choice
> The mp3's that i'm serving were ripped i believe were ripped at 256bps.
> Any help appreciated.
sample rate is the number of audio samples per second. This is usually
44100 for CD quality sound, or sometimes 48000.
bitrate is the number of bits/sec in the stream /after/ it's been
compressed. I.e. how much of your upload bandwidth is being used.
Changing the sample rate affects the quality of the raw, uncompressed
audio file/stream (e.g. a wav). bitrate refers to the compressed stream.
I.e., a 44100 samples/sec 32 kbps stream and a 44100 samples/sec 128 kbps
stream will both decompress to a 44100 samples/sec raw stream, but a 22050
samples/sec 32 kbps stream will decompress to a 22050 samples/sec raw
I think the usual thing to do when streaming is leave the sample rate at
CD quality, and just bring down the bitrate to reduce the bandwidth. If
you're using very low bitrates, you might get better results by
downsampling to a lower sample rate.
If you want to reduce bandwidth, you could have a look at peer to peer
streamers like peercast or streamer (google on the names plus 'peer to
peer broadcast'). With these, every listener is also a relay for other
listeners, so the broadcasting machine only needs enough bandwidth for 2
or 3 streams.
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