interleave/deinterleave in Gst.Bin for gstwebrtc (Python)

[Jack]

I forgot my configuration :
Ubuntu 18.04
GStreamer 1.15.1
++

Jack



Le 25/01/2019 à 15:19, Jack a écrit :
> Hello,
> 
> I have this pipeline encapsulated in a Gst.Bin object in Python working
> gracefully with gstwebrtc :
> 
> filesrc location=17seconds48000D.wav !
> wavparse !
> volume name="vol0" volume=0.5 !
> opusenc frame-size=2 max-payload-size=400 !
> rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> !
> queue
> 
> 
> But this second pipeline doesn't work :
> 
> filesrc location=17seconds48000D.wav !
> wavparse !
> deinterleave name=d
> d.src_0 !
> queue !
> volume name="vol0" volume=0.5 ! i.
> interleave name=i !
> opusenc frame-size=2 max-payload-size=400 !
> rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> !
> queue
> 
> 
> Webrtc seems to work properly but I don't get sound.
> 
> I tested the second pipeline with gst-launch-1.0 and it works fine (no
> error) :
> $ gst-launch-1.0 filesrc location=17seconds48000D.wav ! wavparse !
> deinterleave name=d d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.
> interleave name=i ! opusenc frame-size=2 max-payload-size=400 !
> rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! fakesink
> 
> and if I replace "rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! fakesink" with "oggmux ! filesink location=output.ogg" in this command
> line, the soundfile is good !
> 
> 
> So, my question is : is there an issue between interleave/deinterleave
> and gstwebrtc (or something else) or I missed something ? (bacause I
> couldn't be able to use interleave/deinterleave in a Gst.Bin object ?).
> ++
> 
> Jack
> 
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