I am trying to get around a certain gst bug so I can finish my project and am hoping that an audio expert can point me in the right direction. My pipeline is pretty straight forward at this point:
gst-launch -v -m decklinksrc mode=11 ! alsasink Does anybody know of a way to get around this bug? I have tried quite a few combos of audioconvert but nothing seems to work. Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Got message #4 from element "alsasink0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_NULL, new-state=(GstState)GST_STATE_READY, pending-state=(GstState)GST_STATE_VOID_PENDING; Got message #5 from element "decklinksrc0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_NULL, new-state=(GstState)GST_STATE_READY, pending-state=(GstState)GST_STATE_VOID_PENDING; Got message #6 from element "pipeline0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_NULL, new-state=(GstState)GST_STATE_READY, pending-state=(GstState)GST_STATE_PAUSED; Got message #9 from element "decklinksrc0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_READY, new-state=(GstState)GST_STATE_PAUSED, pending-state=(GstState)GST_STATE_VOID_PENDING; Got message #10 from element "pipeline0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_READY, new-state=(GstState)GST_STATE_PAUSED, pending-state=(GstState)GST_STATE_VOID_PENDING; Setting pipeline to PLAYING ... Got message #11 from element "pipeline0" (new-clock): GstMessageNewClock, clock=(GstClock)"\(GstSystemClock\)\ GstSystemClock"; New clock: GstSystemClock Got message #12 from element "decklinksrc0" (state-changed): GstMessageState, old-state=(GstState)GST_STATE_PAUSED, new-state=(GstState)GST_STATE_PLAYING, pending-state=(GstState)GST_STATE_VOID_PENDING; /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0.GstPad:videosrc: caps = video/x-raw-yuv, format=(fourcc)UYVY, width=(int)1920, height=(int)1080, framerate=(fraction)30000/1001, interlaced=(boolean)true, pixel-aspect-ratio=(fraction)1/1, color-matrix=(string)hdtv, chroma-site=(string)mpeg2 Got message #14 from element "decklinksrc0" (error): /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0.GstPad:audiosrc: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, depth=(int)16, width=(int)16, channels=(int)2, rate=(int)48000 GstMessageError, gerror=(GError)NULL, debug=(string)"gstdecklinksrc.cpp\(1248\):\ gst_decklink_src_task\ \(\):\ /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0"; ERROR: from element /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstdecklinksrc.cpp(1248): gst_decklink_src_task (): /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0 Execution ended after 1002079593 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, depth=(int)16, width=(int)16, channels=(int)2, rate=(int)48000 /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0.GstPad:videosrc: caps = NULL /GstPipeline:pipeline0/GstDecklinkSrc:decklinksrc0.GstPad:audiosrc: caps = NULL Setting pipeline to NULL ... Freeing pipeline ... _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi.
Lawrence Anthony
Are you using SDI input? Can you watch video with "gst-launch decklinksrc mode=11 ! ffmpegcolorspace ! autovideosink"?
In my Intensity Pro card I can not get audio if I am not getting video too[1]. You may use a video fakesink [2].
[1] gst-launch-0.10 decklinksrc mode=720p5994 connection=component audio-input=0 subdevice=1 name=a ! alsasink sync=false a.! queue ! ffmpegcolorspace ! autovideosink sync=false
[2] gst-launch-0.10 decklinksrc mode=720p5994 connection=component audio-input=0 subdevice=1 name=a ! alsasink sync=false a.! queue ! ffmpegcolorspace ! fakesink 2012/9/14 Lawrence Anthony <[hidden email]> I am trying to get around a certain gst bug so I can finish my project and am hoping that an audio expert can point me in the right direction. My pipeline is pretty straight forward at this point: Un Saludo Ruben Gonzalez Gonzalez _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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