interleave/deinterleave in Gst.Bin for gstwebrtc (Python)
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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 _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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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 > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Forget my previous message.
I use now Gst.parse_launch() to put my pipeline and it works nicely. Sorry for the noise. ++ Jack Le 25/01/2019 à 15:34, Jack a écrit : > 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 >> >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel >> > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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