c++ - gstreamer choose one channel and convert to mono (deinterleave) -


i'm creating audio stream audio server , streaming receiver , receiver choose 1 channel , convert mono.

the code below pipeline of receiver. receiving rtp stream. 

gst-launch-0.10 -v \ udpsrc multicast-group=224.0.0.7 port=5000 \ ! "application/x-rtp,media=audio, clock-rate=44100, width=16, height=16, encoding-name=l16, encoding-params=2, payload=96" \ ! gstrtpjitterbuffer latency=200 ! rtpl16depay ! audioconvert ! deinterleave name=d    interleave name=i ! alsasink \ d.src_0 ! queue ! audioconvert !"audio/x-raw-int,channels=1"! i.sink1 \ d.src_0 ! queue ! audioconvert !"audio/x-raw-int,channels=1"! i.sink0

it run listening nothing when stream comes through, throws error. 

error: element /gstpipeline:pipeline0/gstudpsrc:udpsrc0: internal data flow error.

you can't connect d.src_0 twice.

instead, can ditch interleave part , concentrate on playing mono stream (i hope it's ok replaced udpsrc uribin, pipeline easier read way):

gst-launch-0.10 -v uridecodebin uri=file://file.ogg ! audioconvert ! \   deinterleave name=d  d.src0 ! queue ! audioconvert ! alsasink

this will, pipeline, deinterleave stereo stream , take left channel (d.src0, use d.src1 other one) provides mono stream , feed alsasink.

alsa doesn't mind whether feed mono or stereo data it, if explicitly wanted stereo stream/file? add audiopanorama. takes audio stream , places somewhere between left , right speaker (the panorama parameter defaults 0 center) , produces stereo stream:

gst-launch-0.10 -v uridecodebin uri=file://file.mp3 ! audioconvert ! \   deinterleave name=d d.src0 ! queue ! audioconvert ! audiopanorama ! alsasink

but said, alsa (and other sound servers pulse, osd, ...) doesn't care whether feed mono or stereo it, it's gonna play on both channels.


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