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authorMatthias P. Braendli <matthias.braendli@mpb.li>2018-07-16 07:34:25 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2018-07-16 07:34:25 +0200
commita7825f19f14e149f6b14107124ee1521417f2fec (patch)
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Add scenario for LiveWire input to README
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@@ -157,6 +157,31 @@ name for the encoder:
The samplerate of the JACK server should be 32kHz or 48kHz.
+Scenario *LiveWire* or *AES67*
+------------------------------
+
+When audio data is available on the network as a multicast stream, it can be encoded using the following pipeline:
+
+ rtpdump -F payload 239.192.1.1/5004 | \
+ sox -t raw -e signed-integer -r 48000 -c 2 -b 24 -B /dev/stdin -t raw --no-dither -r 48000 -c 2 -b 16 -L /dev/stdout gain 4 | \
+ odr-audioenc -f raw -b $BITRATE -i /dev/stdin -o $DST
+
+It is also possible to use the libvlc input, where you need to create an SDP file with the following contents:
+
+ v=0
+ o=Node 1 1 IN IP4 172.16.235.155
+ s=TestSine
+ t=0 0
+ a=type:multicast
+ c=IN IP4 239.192.0.1
+ m=audio 5004 RTP/AVP 97
+ a=rtpmap:97 L24/48000/2
+
+Replace the IP address in the `o=` field by the one corresponding to your
+source node IP address, and the IP in `c=` by the multicast IP of your stream.
+Then use this SDP file as input for the VLC input.
+
+
Scenario *local file through snd-aloop*
---------------------------------------
Play some local audio source from a file, with ZMQ output for ODR-DabMux. The problem with