hard to decode video to decklink output
Ray Tiley
raytiley at gmail.com
Tue Apr 2 17:13:32 UTC 2019
On Tue, Apr 2, 2019 at 1:05 PM Federico Allegretti <allegfede at gmail.com>
wrote:
> hello, searching the net, seeking a way to playback my digital video files
> out of a black magic video card (i got an intensity pro), i found this
> pipeline to work for most of my video files:
> gst-launch-1.0 uridecodebin
> uri=file:///mnt/nas/postion/of/th/file.extenision name=decode decode. !
> interlace field-pattern=1 ! videoconvert ! video/x-raw,format=UYVY !
> videoscale ! video/x-raw,width=720,height=576 ! videorate !
> video/x-raw,framerate=25/1 ! decklinkvideosink mode=3 decode. !
> audioconvert ! decklinkaudiosink
>
> PS: mode=3 stands for PAL SD 50i as i whant the card to play content out
> of the composite cideo connector.
>
> This works quite well, but with some file playback do not occours.
>
> for example, with a file shot by cell phone i got no problem (those are
> stats by ffmpeg):
> h264 (Baseline) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 13976
> kb/s, SAR 1:1 DAR 16:9, 15.03 fps, 15 tbr, 90k tbn, 180k tbc (default)
> Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 156 kb/s
> (default)
>
> but with another edited with pinnacle studio (stats by ffmpeg):
> Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR
> 1:1 DAR 16:9], 2024 kb/s, 25 fps, 25 tbr, 25k tbn, 50 tbc (default)
> Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
> (default)
>
> i got this error:
>
> ERRORE: dall'elemento
> /GstPipeline:pipeline0/GstURIDecodeBin:decode/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0:
> Internal data stream error.
> Informazioni di debug aggiuntive:
> qtdemux.c(6073): gst_qtdemux_loop ():
> /GstPipeline:pipeline0/GstURIDecodeBin:decode/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0:
> streaming stopped, reason not-negotiated (-4)
> ERRORE: la pipeline non vuole fare il preroll.
> Impostazione della pipeline a NULL ...
> Esecuzione di free sulla pipeline...
>
> I suppose that the problem could be the samplerate, so i resampled the
> audio in this way:
> ffmpeg -i grand_prix_casagrande.mp4 -vcodec copy -acodec aac -ar 48000 -ab
> 128000 -ac 2 grand_prix_casagrande_48.mp4
>
> and now the new file could be played back.
>
> My question now is: There is a way to make the resample AT RUNTIME (and
> maybe only if needed)?
>
> Thanks a lot,
>
> Federico
>
Add an audioresample to your pipeline
decode. ! audioresample ! audioconvert ! decklinkaudiosink
-ray
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