Pipeline not working with gstreamer 1.14.4 or 1.15.1 but is ok with 1.8.3
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Hello !
This is my first message on this list. I woud like to congratuate the developer team for this nice stream framework ! Well, I have a problem with this pipeline with recent version of gstreamer, here the pipeline : ### PIPELINE ################################################ interleave name=i ! opusenc frame-size=20 max-payload-size=4000 ! rtpopuspay pt=96 ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 ! udpsink host=127.0.0.1 port=9999 multifilesrc name="source0" loop=true location="localisation/bruxelles/sounds/output.wav" ! wavparse ignore-length=1 ! decodebin ! deinterleave name=d d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! audiopanorama name="pan0" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d0 d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! audiopanorama name="pan1" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d1 d.src_2 ! audioconvert ! volume name="vol2" volume=0.5 ! audioconvert ! audiopanorama name="pan2" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d2 d.src_3 ! audioconvert ! volume name="vol3" volume=0.5 ! audioconvert ! audiopanorama name="pan3" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d3 d0.src_0 ! audiomixer name=mix0 ! i.sink_0 d0.src_1 ! audiomixer name=mix1 ! i.sink_1 d1.src_0 ! mix0. d1.src_1 ! mix1. d2.src_0 ! mix0. d2.src_1 ! mix1. d3.src_0 ! mix0. d3.src_1 ! mix1. ### END PIPELINE ################################################ with gstreamer 1.14.4 and 1.15.1, I get an error : Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Redistribute latency... ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: Internal data stream error. Additional debug info: gstbasesrc.c(3055): gst_base_src_loop (): /GstPipeline:pipeline0/GstMultiFileSrc:source0: streaming stopped, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... But, with gstreamer 1.8.3, everything is ok : Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Redistribute latency... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock any ideas how to solve this issue ? Thanx. ++ Jack _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hi Jack ! could you try to reduce the pipeline to something shorter that reproduces the issue ? could you also set the GST_DEBUG environment variable to 6 or 7 to get more info about what is wrong ? you could set it before starting you pipeline with something like : GST_DEBUG="7" gst-launch-1.0 .... level 7 is very verbose and you should filter message with something like : GST_DEBUG=audio*:7 (or something else depending on where the issue is, hence my first advice :-)). cheers antoine Le lun. 21 janv. 2019 à 19:27, Jack <[hidden email]> a écrit : Hello ! _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hi Antoine !
Weird, I just try two "smaller" pipelines with a small difference between them : ### PIPELINE 1 ################################################ interleave name=i ! opusenc frame-size=20 max-payload-size=4000 ! rtpopuspay pt=96 ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 ! udpsink host=127.0.0.1 port=9999 multifilesrc name="source0" loop=true location="localisation/bruxelles/sounds/output.wav" ! wavparse ignore-length=1 ! decodebin ! deinterleave name=d d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! audiopanorama name="pan0" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d0 d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! audiopanorama name="pan1" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d1 d0.src_0 ! audiomixer name=mix0 ! i.sink_0 d0.src_1 ! audiomixer name=mix1 ! i.sink_1 ### END PIPELINE 1 ################################################ ### PIPELINE 2 ################################################ interleave name=i ! opusenc frame-size=20 max-payload-size=4000 ! rtpopuspay pt=96 ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 ! udpsink host=127.0.0.1 port=9999 multifilesrc name="source0" loop=true location="localisation/bruxelles/sounds/output.wav" ! wavparse ignore-length=1 ! decodebin ! deinterleave name=d d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! audiopanorama name="pan0" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d0 d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! audiopanorama name="pan1" panorama=0 ! audioconvert ! audio/x-raw,channels=2 ! deinterleave name=d1 d0.src_0 ! audiomixer name=mix0 ! i.sink_0 ### END PIPELINE 2 ################################################ The Pipeline 1 doesn't work and the pipeline 2 is OK. The only difference between then is the last line in the pipeline 1 (which get the second channel of d0 and ) : d0.src_1 ! audiomixer name=mix1 ! i.sink_1 Maybe that can help to solve this issue ? OTOH, I tried with "GST_DEBUG=audio*:7" and didn't get red lines, can I say that all is OK with audio ? ++ Jack Le 21/01/2019 à 21:52, Antoine Villeret a écrit : > Hi Jack ! > > could you try to reduce the pipeline to something shorter that > reproduces the issue ? > could you also set the GST_DEBUG environment variable to 6 or 7 to get > more info about what is wrong ? > > you could set it before starting you pipeline with something like : > > GST_DEBUG="7" gst-launch-1.0 .... > > level 7 is very verbose and you should filter message with something > like : GST_DEBUG=audio*:7 (or something else depending on where the > issue is, hence my first advice :-)). > > cheers > > antoine > > Le lun. 21 janv. 2019 à 19:27, Jack <[hidden email] > <mailto:[hidden email]>> a écrit : > > Hello ! > > This is my first message on this list. > I woud like to congratuate the developer team for this nice stream > framework ! > > Well, I have a problem with this pipeline with recent version of > gstreamer, here the pipeline : > > ### PIPELINE ################################################ > interleave name=i ! > opusenc frame-size=20 max-payload-size=4000 ! > rtpopuspay pt=96 ! > application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 > ! > udpsink host=127.0.0.1 port=9999 > > multifilesrc name="source0" loop=true > location="localisation/bruxelles/sounds/output.wav" ! > wavparse ignore-length=1 ! > decodebin ! deinterleave name=d > > d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! > audiopanorama name="pan0" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d0 > d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! > audiopanorama name="pan1" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d1 > d.src_2 ! audioconvert ! volume name="vol2" volume=0.5 ! audioconvert ! > audiopanorama name="pan2" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d2 > d.src_3 ! audioconvert ! volume name="vol3" volume=0.5 ! audioconvert ! > audiopanorama name="pan3" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d3 > > d0.src_0 ! audiomixer name=mix0 ! i.sink_0 > d0.src_1 ! audiomixer name=mix1 ! i.sink_1 > d1.src_0 ! mix0. > d1.src_1 ! mix1. > d2.src_0 ! mix0. > d2.src_1 ! mix1. > d3.src_0 ! mix0. > d3.src_1 ! mix1. > ### END PIPELINE ################################################ > > with gstreamer 1.14.4 and 1.15.1, I get an error : > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Redistribute latency... > ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: > Internal data stream error. > Additional debug info: > gstbasesrc.c(3055): gst_base_src_loop (): > /GstPipeline:pipeline0/GstMultiFileSrc:source0: > streaming stopped, reason not-negotiated (-4) > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > But, with gstreamer 1.8.3, everything is ok : > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Redistribute latency... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > > any ideas how to solve this issue ? > Thanx. > ++ > > Jack > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > <mailto:[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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Need help...
I just try this pipeline, but it still fails with Gstreamer 1.15.1 (it works perfectly with Gstreamer 1.8.3 !) : ### PIPELINE ################################################ multifilesrc name="source0" loop=true location=output.wav ! wavparse ignore-length=1 ! deinterleave name=d d.src_0 ! volume name="vol0" volume=0.5 ! audiopanorama name="pan0" panorama=0 ! audio/x-raw,channels=2 ! deinterleave name=d0 d.src_1 ! volume name="vol1" volume=0.5 ! audiopanorama name="pan1" panorama=0 ! audio/x-raw,channels=2 ! deinterleave name=d1 audiomixer name=mix0 ! i.sink_0 audiomixer name=mix1 ! i.sink_1 d0.src_0 ! mix0. d0.src_1 ! mix1. d1.src_0 ! mix0. d1.src_1 ! mix1. interleave name=i ! opusenc frame-size=20 max-payload-size=4000 ! rtpopuspay pt=96 ! fakesink ### END PIPELINE ################################################ With GST_DEBUG=3 I get : Setting pipeline to PAUSED ... Pipeline is PREROLLING ... 0:00:00.198717531 30503 0x55a787f7e940 WARN interleave interleave.c:316:gst_interleave_set_channel_positions:<i> Invalid channel positions, using NONE 0:00:00.198774934 30503 0x55a787f7e940 FIXME basesink gstbasesink.c:3248:gst_base_sink_default_event:<fakesink0> stream-start event without group-id. Consider implementing group-id handling in the upstream elements Redistribute latency... 0:00:00.199571368 30503 0x55a787f7e940 WARN GST_PADS gstpad.c:4230:gst_pad_peer_query:<opusenc0:src> could not send sticky events 0:00:00.362518006 30503 0x55a787f7ea30 WARN basesrc gstbasesrc.c:3064:gst_base_src_loop:<source0> error: Internal data stream error. 0:00:00.362538938 30503 0x55a787f7ea30 WARN basesrc gstbasesrc.c:3064:gst_base_src_loop:<source0> error: streaming stopped, reason not-negotiated (-4) ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: Internal data stream error. Additional debug info: gstbasesrc.c(3064): gst_base_src_loop (): /GstPipeline:pipeline0/GstMultiFileSrc:source0: streaming stopped, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... If i replace "rtpopuspay pt=96 ! fakesink" with "oggmux ! filesink location=test.ogg", the pipeline is OK. It also works if I remove : "audiomixer name=mix1 ! i.sink_1" "d0.src_1 ! mix1." "d1.src_1 ! mix1." I don't know what to try now, if someone has an idee ? ++ Jack Le 22/01/2019 à 11:16, Jack a écrit : > Hi Antoine ! > > Weird, I just try two "smaller" pipelines with a small difference > between them : > > ### PIPELINE 1 ################################################ > interleave name=i ! > opusenc frame-size=20 max-payload-size=4000 ! > rtpopuspay pt=96 ! > application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 > ! > udpsink host=127.0.0.1 port=9999 > > multifilesrc name="source0" loop=true > location="localisation/bruxelles/sounds/output.wav" ! > wavparse ignore-length=1 ! > decodebin ! > deinterleave name=d > > d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! > audiopanorama name="pan0" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d0 > d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! > audiopanorama name="pan1" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d1 > > d0.src_0 ! audiomixer name=mix0 ! i.sink_0 > d0.src_1 ! audiomixer name=mix1 ! i.sink_1 > ### END PIPELINE 1 ################################################ > > ### PIPELINE 2 ################################################ > interleave name=i ! > opusenc frame-size=20 max-payload-size=4000 ! > rtpopuspay pt=96 ! > application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 > ! > udpsink host=127.0.0.1 port=9999 > > multifilesrc name="source0" loop=true > location="localisation/bruxelles/sounds/output.wav" ! > wavparse ignore-length=1 ! > decodebin ! > deinterleave name=d > > d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! > audiopanorama name="pan0" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d0 > d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! > audiopanorama name="pan1" panorama=0 ! audioconvert ! > audio/x-raw,channels=2 ! deinterleave name=d1 > > d0.src_0 ! audiomixer name=mix0 ! i.sink_0 > ### END PIPELINE 2 ################################################ > > The Pipeline 1 doesn't work and the pipeline 2 is OK. > The only difference between then is the last line in the pipeline 1 > (which get the second channel of d0 and ) : > d0.src_1 ! audiomixer name=mix1 ! i.sink_1 > > Maybe that can help to solve this issue ? > > OTOH, I tried with "GST_DEBUG=audio*:7" and didn't get red lines, can I > say that all is OK with audio ? > ++ > > Jack > > > > Le 21/01/2019 à 21:52, Antoine Villeret a écrit : >> Hi Jack ! >> >> could you try to reduce the pipeline to something shorter that >> reproduces the issue ? >> could you also set the GST_DEBUG environment variable to 6 or 7 to get >> more info about what is wrong ? >> >> you could set it before starting you pipeline with something like : >> >> GST_DEBUG="7" gst-launch-1.0 .... >> >> level 7 is very verbose and you should filter message with something >> like : GST_DEBUG=audio*:7 (or something else depending on where the >> issue is, hence my first advice :-)). >> >> cheers >> >> antoine >> >> Le lun. 21 janv. 2019 à 19:27, Jack <[hidden email] >> <mailto:[hidden email]>> a écrit : >> >> Hello ! >> >> This is my first message on this list. >> I woud like to congratuate the developer team for this nice stream >> framework ! >> >> Well, I have a problem with this pipeline with recent version of >> gstreamer, here the pipeline : >> >> ### PIPELINE ################################################ >> interleave name=i ! >> opusenc frame-size=20 max-payload-size=4000 ! >> rtpopuspay pt=96 ! >> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 >> ! >> udpsink host=127.0.0.1 port=9999 >> >> multifilesrc name="source0" loop=true >> location="localisation/bruxelles/sounds/output.wav" ! >> wavparse ignore-length=1 ! >> decodebin ! deinterleave name=d >> >> d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! >> audiopanorama name="pan0" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d0 >> d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! >> audiopanorama name="pan1" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d1 >> d.src_2 ! audioconvert ! volume name="vol2" volume=0.5 ! audioconvert ! >> audiopanorama name="pan2" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d2 >> d.src_3 ! audioconvert ! volume name="vol3" volume=0.5 ! audioconvert ! >> audiopanorama name="pan3" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d3 >> >> d0.src_0 ! audiomixer name=mix0 ! i.sink_0 >> d0.src_1 ! audiomixer name=mix1 ! i.sink_1 >> d1.src_0 ! mix0. >> d1.src_1 ! mix1. >> d2.src_0 ! mix0. >> d2.src_1 ! mix1. >> d3.src_0 ! mix0. >> d3.src_1 ! mix1. >> ### END PIPELINE ################################################ >> >> with gstreamer 1.14.4 and 1.15.1, I get an error : >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> Redistribute latency... >> ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: >> Internal data stream error. >> Additional debug info: >> gstbasesrc.c(3055): gst_base_src_loop (): >> /GstPipeline:pipeline0/GstMultiFileSrc:source0: >> streaming stopped, reason not-negotiated (-4) >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> But, with gstreamer 1.8.3, everything is ok : >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> Redistribute latency... >> Pipeline is PREROLLED ... >> Setting pipeline to PLAYING ... >> New clock: GstSystemClock >> >> any ideas how to solve this issue ? >> Thanx. >> ++ >> >> Jack >> >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> <mailto:[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 > _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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Hello,
I have a simpler pipeline, but still have the issue (with v1.15.1, not with v1.8.3 so it is a bug ?) : GST_DEBUG=3 gst-launch-1.0 filesrc location=output.wav ! wavparse ! deinterleave name=d d.src_0 ! volume name="vol0" volume=0.5 ! audiopanorama name="pan0" panorama=0 ! audio/x-raw,channels=2 ! audioconvert ! queue ! i. d.src_1 ! volume name="vol1" volume=0.5 ! audiopanorama name="pan1" panorama=0 ! audio/x-raw,channels=2 ! audioconvert ! queue ! i. interleave name=i ! opusenc frame-size=20 max-payload-size=4000 ! rtpopuspay pt=96 ! fakesink I get : Setting pipeline to PAUSED ... 0:00:00.014909281 25566 0x5569f3678550 WARN basesrc gstbasesrc.c:3592:gst_base_src_start_complete:<filesrc0> pad not activated yet Pipeline is PREROLLING ... 0:00:00.015098902 25566 0x5569f3668de0 FIXME default gstutils.c:3981:gst_pad_create_stream_id_internal:<wavparse0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id 0:00:00.020775475 25566 0x5569f3668c50 WARN interleave interleave.c:316:gst_interleave_set_channel_positions:<i> Invalid channel positions, using NONE 0:00:00.020796839 25566 0x5569f3668c50 FIXME basesink gstbasesink.c:3248:gst_base_sink_default_event:<fakesink0> stream-start event without group-id. Consider implementing group-id handling in the upstream elements 0:00:00.020936961 25566 0x5569f3668c50 FIXME interleave interleave.c:920:gst_interleave_sink_event:<i> FIXME: merge tags and send after stream-start 0:00:00.021016249 25566 0x5569f3668c00 FIXME interleave interleave.c:920:gst_interleave_sink_event:<i> FIXME: merge tags and send after stream-start 0:00:00.021754444 25566 0x5569f3668c50 WARN GST_PADS gstpad.c:4230:gst_pad_peer_query:<opusenc0:src> could not send sticky events Redistribute latency... 0:00:00.023276294 25566 0x5569f3668de0 WARN wavparse gstwavparse.c:2297:gst_wavparse_loop:<wavparse0> error: Internal data stream error. 0:00:00.023285254 25566 0x5569f3668de0 WARN wavparse gstwavparse.c:2297:gst_wavparse_loop:<wavparse0> error: streaming stopped, reason not-negotiated (-4) 0:00:00.023313614 25566 0x5569f3668de0 WARN queue gstqueue.c:988:gst_queue_handle_sink_event:<queue1> error: Internal data stream error. 0:00:00.023321091 25566 0x5569f3668de0 WARN queue gstqueue.c:988:gst_queue_handle_sink_event:<queue1> error: streaming stopped, reason not-negotiated (-4) ERROR: from element /GstPipeline:pipeline0/GstWavParse:wavparse0: Internal data stream error. Additional debug info: gstwavparse.c(2297): gst_wavparse_loop (): /GstPipeline:pipeline0/GstWavParse:wavparse0: streaming stopped, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... The soundfile "output.wav" is : Input #0, wav, from 'localisation/bruxelles/sounds/output.wav': Metadata: date : 2018-06-20 encoder : Lavf56.40.101 encoded_by : REAPER Duration: 00:18:40.00, bitrate: 3072 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 4.0, s16, 3072 kb/s Any help to solve this issue would be very appreciated ! ++ Jack Le 23/01/2019 à 00:53, Jack a écrit : > Need help... > > I just try this pipeline, but it still fails with Gstreamer 1.15.1 (it > works perfectly with Gstreamer 1.8.3 !) : > > ### PIPELINE ################################################ > multifilesrc name="source0" loop=true location=output.wav ! > wavparse ignore-length=1 ! > deinterleave name=d > > d.src_0 ! volume name="vol0" volume=0.5 ! audiopanorama name="pan0" > panorama=0 ! audio/x-raw,channels=2 ! deinterleave name=d0 > d.src_1 ! volume name="vol1" volume=0.5 ! audiopanorama name="pan1" > panorama=0 ! audio/x-raw,channels=2 ! deinterleave name=d1 > > audiomixer name=mix0 ! i.sink_0 > audiomixer name=mix1 ! i.sink_1 > > d0.src_0 ! mix0. > d0.src_1 ! mix1. > d1.src_0 ! mix0. > d1.src_1 ! mix1. > > interleave name=i ! > opusenc frame-size=20 max-payload-size=4000 ! > rtpopuspay pt=96 ! > fakesink > ### END PIPELINE ################################################ > > > With GST_DEBUG=3 I get : > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > 0:00:00.198717531 30503 0x55a787f7e940 WARN interleave > interleave.c:316:gst_interleave_set_channel_positions:<i> Invalid > channel positions, using NONE > 0:00:00.198774934 30503 0x55a787f7e940 FIXME basesink > gstbasesink.c:3248:gst_base_sink_default_event:<fakesink0> stream-start > event without group-id. Consider implementing group-id handling in the > upstream elements > Redistribute latency... > 0:00:00.199571368 30503 0x55a787f7e940 WARN GST_PADS > gstpad.c:4230:gst_pad_peer_query:<opusenc0:src> could not send sticky events > 0:00:00.362518006 30503 0x55a787f7ea30 WARN basesrc > gstbasesrc.c:3064:gst_base_src_loop:<source0> error: Internal data > stream error. > 0:00:00.362538938 30503 0x55a787f7ea30 WARN basesrc > gstbasesrc.c:3064:gst_base_src_loop:<source0> error: streaming stopped, > reason not-negotiated (-4) > ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: > Internal data stream error. > Additional debug info: > gstbasesrc.c(3064): gst_base_src_loop (): > /GstPipeline:pipeline0/GstMultiFileSrc:source0: > streaming stopped, reason not-negotiated (-4) > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > > If i replace "rtpopuspay pt=96 ! fakesink" with "oggmux ! filesink > location=test.ogg", the pipeline is OK. > > It also works if I remove : > "audiomixer name=mix1 ! i.sink_1" > "d0.src_1 ! mix1." > "d1.src_1 ! mix1." > > I don't know what to try now, if someone has an idee ? > ++ > > Jack > > > > > Le 22/01/2019 à 11:16, Jack a écrit : >> Hi Antoine ! >> >> Weird, I just try two "smaller" pipelines with a small difference >> between them : >> >> ### PIPELINE 1 ################################################ >> interleave name=i ! >> opusenc frame-size=20 max-payload-size=4000 ! >> rtpopuspay pt=96 ! >> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 >> ! >> udpsink host=127.0.0.1 port=9999 >> >> multifilesrc name="source0" loop=true >> location="localisation/bruxelles/sounds/output.wav" ! >> wavparse ignore-length=1 ! >> decodebin ! >> deinterleave name=d >> >> d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! >> audiopanorama name="pan0" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d0 >> d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! >> audiopanorama name="pan1" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d1 >> >> d0.src_0 ! audiomixer name=mix0 ! i.sink_0 >> d0.src_1 ! audiomixer name=mix1 ! i.sink_1 >> ### END PIPELINE 1 ################################################ >> >> ### PIPELINE 2 ################################################ >> interleave name=i ! >> opusenc frame-size=20 max-payload-size=4000 ! >> rtpopuspay pt=96 ! >> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 >> ! >> udpsink host=127.0.0.1 port=9999 >> >> multifilesrc name="source0" loop=true >> location="localisation/bruxelles/sounds/output.wav" ! >> wavparse ignore-length=1 ! >> decodebin ! >> deinterleave name=d >> >> d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! >> audiopanorama name="pan0" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d0 >> d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! >> audiopanorama name="pan1" panorama=0 ! audioconvert ! >> audio/x-raw,channels=2 ! deinterleave name=d1 >> >> d0.src_0 ! audiomixer name=mix0 ! i.sink_0 >> ### END PIPELINE 2 ################################################ >> >> The Pipeline 1 doesn't work and the pipeline 2 is OK. >> The only difference between then is the last line in the pipeline 1 >> (which get the second channel of d0 and ) : >> d0.src_1 ! audiomixer name=mix1 ! i.sink_1 >> >> Maybe that can help to solve this issue ? >> >> OTOH, I tried with "GST_DEBUG=audio*:7" and didn't get red lines, can I >> say that all is OK with audio ? >> ++ >> >> Jack >> >> >> >> Le 21/01/2019 à 21:52, Antoine Villeret a écrit : >>> Hi Jack ! >>> >>> could you try to reduce the pipeline to something shorter that >>> reproduces the issue ? >>> could you also set the GST_DEBUG environment variable to 6 or 7 to get >>> more info about what is wrong ? >>> >>> you could set it before starting you pipeline with something like : >>> >>> GST_DEBUG="7" gst-launch-1.0 .... >>> >>> level 7 is very verbose and you should filter message with something >>> like : GST_DEBUG=audio*:7 (or something else depending on where the >>> issue is, hence my first advice :-)). >>> >>> cheers >>> >>> antoine >>> >>> Le lun. 21 janv. 2019 à 19:27, Jack <[hidden email] >>> <mailto:[hidden email]>> a écrit : >>> >>> Hello ! >>> >>> This is my first message on this list. >>> I woud like to congratuate the developer team for this nice stream >>> framework ! >>> >>> Well, I have a problem with this pipeline with recent version of >>> gstreamer, here the pipeline : >>> >>> ### PIPELINE ################################################ >>> interleave name=i ! >>> opusenc frame-size=20 max-payload-size=4000 ! >>> rtpopuspay pt=96 ! >>> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000 >>> ! >>> udpsink host=127.0.0.1 port=9999 >>> >>> multifilesrc name="source0" loop=true >>> location="localisation/bruxelles/sounds/output.wav" ! >>> wavparse ignore-length=1 ! >>> decodebin ! deinterleave name=d >>> >>> d.src_0 ! audioconvert ! volume name="vol0" volume=0.5 ! audioconvert ! >>> audiopanorama name="pan0" panorama=0 ! audioconvert ! >>> audio/x-raw,channels=2 ! deinterleave name=d0 >>> d.src_1 ! audioconvert ! volume name="vol1" volume=0.5 ! audioconvert ! >>> audiopanorama name="pan1" panorama=0 ! audioconvert ! >>> audio/x-raw,channels=2 ! deinterleave name=d1 >>> d.src_2 ! audioconvert ! volume name="vol2" volume=0.5 ! audioconvert ! >>> audiopanorama name="pan2" panorama=0 ! audioconvert ! >>> audio/x-raw,channels=2 ! deinterleave name=d2 >>> d.src_3 ! audioconvert ! volume name="vol3" volume=0.5 ! audioconvert ! >>> audiopanorama name="pan3" panorama=0 ! audioconvert ! >>> audio/x-raw,channels=2 ! deinterleave name=d3 >>> >>> d0.src_0 ! audiomixer name=mix0 ! i.sink_0 >>> d0.src_1 ! audiomixer name=mix1 ! i.sink_1 >>> d1.src_0 ! mix0. >>> d1.src_1 ! mix1. >>> d2.src_0 ! mix0. >>> d2.src_1 ! mix1. >>> d3.src_0 ! mix0. >>> d3.src_1 ! mix1. >>> ### END PIPELINE ################################################ >>> >>> with gstreamer 1.14.4 and 1.15.1, I get an error : >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> Redistribute latency... >>> ERROR: from element /GstPipeline:pipeline0/GstMultiFileSrc:source0: >>> Internal data stream error. >>> Additional debug info: >>> gstbasesrc.c(3055): gst_base_src_loop (): >>> /GstPipeline:pipeline0/GstMultiFileSrc:source0: >>> streaming stopped, reason not-negotiated (-4) >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> But, with gstreamer 1.8.3, everything is ok : >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> Redistribute latency... >>> Pipeline is PREROLLED ... >>> Setting pipeline to PLAYING ... >>> New clock: GstSystemClock >>> >>> any ideas how to solve this issue ? >>> Thanx. >>> ++ >>> >>> Jack >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> <mailto:[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 >> > > _______________________________________________ > 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 |
Re: Pipeline not working with gstreamer 1.14.4 or 1.15.1 but is ok with 1.8.3
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On Thu, 2019-01-24 at 12:20 +0100, Jack wrote:
> Hello, > > I have a simpler pipeline, but still have the issue (with v1.15.1, > not with v1.8.3 so it is a bug ?) : Most likely yes, and most likely it's caused by interleave having being ported to the new audio aggregator base class. Please create an issue here: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/new -- Sebastian Dröge, Centricular Ltd · https://www.centricular.com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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