WAV broadcast over RTP/UDP multicast

[joextodd]

Hi everyone, 

I am trying to use gstreamer to broadcast a WAV file over RTP (UDP
multicast) to a couple of RPis to create a Sonos like music sharing system
for my home. My pipelines seem to work ok for mono, L16 at 44.1kHz, but when
moving to stereo it becomes quite glitchy. I'm assuming this should be
possible since this is designed for audiovisual bitrates. Can anyone see any
issues with my pipeline?

*SENDER*
SRC="wavparse ! audioconvert ! audioresample"
ENC="rtpL16pay ! rtprtxqueue"

gst-launch-1.0 -v rtpbin name=rtpbin ntp-time-source=ntp rtp-profile=avpf \
filesrc location=$1 ! tee name=t ! queue ! $SRC ! $ENC !
rtpbin.send_rtp_sink_0  \
rtpbin.send_rtp_src_0 ! udpsink host=$IP auto-multicast=true port=1223
sync=true async=false \
rtpbin.send_rtcp_src_0 ! udpsink host=$IP auto-multicast=true port=1224
sync=false async=false \
udpsrc address=$IP auto-multicast=true port=1225 ! rtpbin.recv_rtcp_sink_0

*RECEIVER*
RTP="rtp-profile=avpf ntp-time-source=ntp ntp-sync=true buffer-mode=synced
do-timestamp=true do-retransmission=true do-lost=true latency=1000
rtcp-sync-send-time=false"
CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)$RATE,encoding-name=(string)L16,channels=(int)$CHAN,payload=(int)96"
DEC="rtpL16depay"
SINK="audiomixer name=audiomix ! alsasink"

gst-launch-1.0 -v rtpbin name=rtpbin $RTP \
udpsrc address=$IP auto-multicast=true port=1223 caps=$CAPS \
! rtpbin.recv_rtp_sink_0 rtpbin. ! $DEC ! $SINK \
udpsrc address=$IP auto-multicast=true port=1224 \
! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink host=$IP
auto-multicast=true port=1225

Any help would be much appreciated.
Thanks



--
Sent from: http://gstreamer-devel.966125.n4.nabble.com/