<div dir="ltr"><div>Hi Patrick,</div><div>I think I figured it out.</div><div>Before switching to a new version, some housekeeping tasks have to be done. This is what I do:</div><div>1. First, I uninstall all the artifacts. So, for each repository (gstreamer, gst-plugins-good, etc.), I issue sudo make uninstall</div><div>2. Then, I run sudo ldconfig.</div><div>3. I clean all the repositories (actually, I erase them and download them again to have a completely fresh checkout ... I'm pretty sure there is a better way to do that, something like "make clean" or similar).</div><div>4. I checkout the tag of interest.</div><div>5. I build and install all the repositories.</div><div> </div><div>This way I successfully moved from 1.16.2 to 1.14.4.</div><div> </div><div>Andrea</div> <div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"> Date: Wed, 29 Apr 2020 07:02:45 -0700 From: Patrick Cusack <<a href="mailto:patrickcusack@mac.com" target="_blank">patrickcusack@mac.com</a>> To: Discussion of the development of and with GStreamer <<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a>> Subject: Re: Building different versions of gstreamer Message-ID: <<a href="mailto:45A75961-0E11-483B-AC99-2ECFF48F0642@mac.com" target="_blank">45A75961-0E11-483B-AC99-2ECFF48F0642@mac.com</a>> Content-Type: text/plain; charset="utf-8" I don?t think Uninstalling is the issue. You are building an older version. Could you have incompatible libraries? Sent from my iPhone > On Apr 29, 2020, at 12:00 AM, Andrea Marson <<a href="mailto:andrea.marson.dave@gmail.com" target="_blank">andrea.marson.dave@gmail.com</a>> wrote: > > ? > Hi, > I'm working with gstreamer on a PC running Ubuntu 18.04. > For my project, I need to build different versions of gstreamer along with its plugins. > For instance, I recently built tag 1.16.2. After that, I needed to build tag 1.14.4. However, building of gst-plugins-base 1.14.4 failed: > > linking of temporary binary failed: Command '['../../../libtool', '--mode=link', '--tag=CC', 'gcc', '-o', '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/pbutils/tmp-introspectvhvale74/GstPbutils-1.0', '-export-dynamic', '-g', '-O2', '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/pbutils/tmp-introspectvhvale74/GstPbutils-1.0.o', '-L.', '<a href="http://libgstpbutils-1.0.la" rel="noreferrer" target="_blank">libgstpbutils-1.0.la</a>', '-L/usr/local/lib', '-rpath', '/usr/local/lib', '-L/usr/local/lib', '-rpath', '/usr/local/lib', '-L../../../gst-libs/gst/tag/', '-L../../../gst-libs/gst/video/', '-L../../../gst-libs/gst/audio/', '-L/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/tag/.libs', '-L/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/video/.libs', '-L/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/audio/.libs', '-L/usr/local/lib', '-lgio-2.0', '-Wl,--export-dynamic', '-lgmodule-2.0', '-p thread', '-lgsttag-1.0', '-lgstvideo-1.0', '-lgstaudio-1.0', '-lgstbase-1.0', '-lgstreamer-1.0', '-lgobject-2.0', '-lglib-2.0']' returned non-zero exit status 1. > Makefile:1285: recipe for target 'GstPbutils-1.0.gir' failed > make[5]: *** [GstPbutils-1.0.gir] Error 1 > make[5]: Leaving directory '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/pbutils' > Makefile:728: recipe for target 'all' failed > make[4]: *** [all] Error 2 > make[4]: Leaving directory '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst/pbutils' > Makefile:659: recipe for target 'all-recursive' failed > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs/gst' > Makefile:624: recipe for target 'all-recursive' failed > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base/gst-libs' > Makefile:748: recipe for target 'all-recursive' failed > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory '/home/sysadmin/devel/hfr-camera/gst/build-gst-1.14/gst-plugins-base' > Makefile:679: recipe for target 'all' failed > make: *** [all] Error 2 > > I suspect this is due to the fact that I did not uninstall gstreamer cleanly before building tag 1.14.4. > How to cleanly uninstall tag 1.16.2, which I installed by issuing "sudo make install"? > > Thank you in advance for you help. > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> > <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20200429/dbb5121a/attachment-0001.htm" rel="noreferrer" target="_blank">https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20200429/dbb5121a/attachment-0001.htm</a>> ------------------------------ Message: 2 Date: Wed, 29 Apr 2020 17:27:38 +0200 From: Juan Navarro <<a href="mailto:juan.navarro@gmx.es" target="_blank">juan.navarro@gmx.es</a>> To: <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> Subject: Constant avdec_h264: decoding error Message-ID: <<a href="mailto:7039839d-3559-5deb-03e0-04cc1a18ea8b@gmx.es" target="_blank">7039839d-3559-5deb-03e0-04cc1a18ea8b@gmx.es</a>> Content-Type: text/plain; charset=utf-8; format=flowed I'm getting a continuous stream of WARN and ERROR messages when trying to play an RTSP source, due to issues in the H.264 decoding done by avdec_h264. This is a (truncated) sample of the messages: 0:00:02.186 WARN? libav gstavcodecmap.c:2441:gst_ffmpeg_caps_to_pixfmt: ignoring insane framerate 1/0 0:00:02.187 ERROR libav :0:: no frame! 0:00:02.187 WARN? libav gstavviddec.c:1535:gst_ffmpegviddec_frame:<avdec_h264-0> avdec_h264: decoding error (len: -1094995529, have_data: 0) 0:00:02.189 ERROR libav :0:: left block unavailable for requested intra mode at 0 2 0:00:02.189 ERROR libav :0:: error while decoding MB 0 2, bytestream 10023 0:00:02.201 ERROR libav :0:: no frame! 0:00:02.201 WARN? libav gstavviddec.c:1535:gst_ffmpegviddec_frame:<avdec_h264-0> avdec_h264: decoding error (len: -1094995529, have_data: 0) 0:00:02.202 ERROR libav :0:: no frame! 0:00:02.202 WARN? libav gstavviddec.c:1535:gst_ffmpegviddec_frame:<avdec_h264-0> avdec_h264: decoding error (len: -1094995529, have_data: 0) 0:00:02.202 ERROR libav :0:: Reference 4 >= 2 0:00:02.202 ERROR libav :0:: error while decoding MB 35 4, bytestream 2615 0:00:02.202 ERROR libav :0:: no frame! 0:00:02.202 WARN? libav gstavviddec.c:1535:gst_ffmpegviddec_frame:<avdec_h264-0> avdec_h264: decoding error (len: -1094995529, have_data: 0) 0:00:02.204 ERROR libav :0:: Reference 2 >= 2 0:00:02.204 ERROR libav :0:: error while decoding MB 29 4, bytestream 3647 0:00:02.209 ERROR libav :0:: no frame! 0:00:02.209 WARN? libav gstavviddec.c:1535:gst_ffmpegviddec_frame:<avdec_h264-0> avdec_h264: decoding error (len: -1094995529, have_data: 0) 0:00:02.209 ERROR libav :0:: Reference 3 >= 2 0:00:02.209 ERROR libav :0:: error while decoding MB 36 10, bytestream 4386 0:00:02.210 ERROR libav :0:: no frame! As you can see from the timestamps, these are printed dozens of times per second. The RTSP stream used to get these errors is this one: rtsp://<a href="http://95.131.167.44:554/10701947" rel="noreferrer" target="_blank">95.131.167.44:554/10701947</a> Hopefully this stays accessible for the next days / weeks. My gst-launch command is like this: gst-launch-1.0 \ ??? rtspsrc location="rtsp://<a href="http://95.131.167.44:554/10701947" rel="noreferrer" target="_blank">95.131.167.44:554/10701947</a>" protocols="tcp" \ ??? ! "application/x-rtp, media=(string)video" \ ??? ! decodebin \ ??? ! autovideosink Of course, a completely manual equivalent has the same issues: gst-launch-1.0 \ ??? rtspsrc location="rtsp://<a href="http://95.131.167.44:554/10701947" rel="noreferrer" target="_blank">95.131.167.44:554/10701947</a>" protocols="tcp" \ ??? ! "application/x-rtp, media=(string)video" \ ??? ! rtph264depay \ ??? ! h264parse \ ??? ! avdec_h264 \ ??? ! autovideosink I have been able to test and got same results with the GStreamer versions available in Ubuntu 16.04 (Gstreamer 1.8.3), 18.04 (GStreamer 1.14.5), and 20.04 (GStreamer 1.16.2). Curious thing is that even despite all those errors and warnings, the end result seems to be playing back fluidly and without issues. Also if audio is added to the pipeline, it also is correct and synchronized with the video. Nevertheless, all those warnings must be indicating some issue that is ingrained in the encoding of the video, and I'd love some help figuring out if there is something a receiver application can do to solve it, or if there might be a bug to be reported in the decoder itself. Thanks! Juan Navarro ------------------------------ Message: 3 Date: Wed, 29 Apr 2020 07:58:45 -0500 (CDT) From: Alex A <<a href="mailto:alex.valgusnov@microimpuls.com" target="_blank">alex.valgusnov@microimpuls.com</a>> To: <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> Subject: GstAggregator doesn't consume buffers Message-ID: <<a href="mailto:1588165125527-0.post@n4.nabble.com" target="_blank">1588165125527-0.post@n4.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hello, I'm trying to write gstreamer plugin for `RGBA bitmap -> DVB Subtitles` encoding. First of all I wrote simple stub: element just emulate encoding by producing static dvb subtitles from file on every input buffer. If run this pipeline: then result is fine: DVB-Inspector sees subtitles and it's valid. But if add another stream to muxer (e.g. audio or video): then the resulting file will not contain subtitles. Even in the Program Map Table there are no entry for it. Logs of `queue` between `dvbsubenc` and `mpegtsmux`: On a src pad, time is only updated for first few buffers. It looks like mpegtsmux (more precisely, GstAggregator) blocks `queue:src` thread and `queue` can't push data. Information about that this stream is sparse, as well as gap events, are supplied by the demuxer (pad with teletext) and just forwarded by other elements to `mpegtsmux`. In GstAggregator logs: "Done chaining" prints only for first few buffers, and then: Line "pad not ready to be consumed yet" are repeated endlessly. And after that, the time on the queue src pad ceases to be updated. Can anyone explain what is happening and what am I doing wrong? Code and file with subtitles: gstdvbsubenc.c <<a href="http://gstreamer-devel.966125.n4.nabble.com/file/t379422/gstdvbsubenc.c" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/file/t379422/gstdvbsubenc.c</a>> gstdvbsubenc.h <<a href="http://gstreamer-devel.966125.n4.nabble.com/file/t379422/gstdvbsubenc.h" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/file/t379422/gstdvbsubenc.h</a>> dvb-dump.bin <<a href="http://gstreamer-devel.966125.n4.nabble.com/file/t379422/dvb-dump.bin" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/file/t379422/dvb-dump.bin</a>> -- Sent from: <a href="http://gstreamer-devel.966125.n4.nabble.com/" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/</a> ------------------------------ Message: 4 Date: Wed, 29 Apr 2020 10:49:29 -0500 (CDT) From: LuddeBoi1337 <<a href="mailto:ludvigthor97@gmail.com" target="_blank">ludvigthor97@gmail.com</a>> To: <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> Subject: x265enc pipeline on GStreamer 1.14.5 Message-ID: <<a href="mailto:1588175369056-0.post@n4.nabble.com" target="_blank">1588175369056-0.post@n4.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hello! I am trying to build a pipeline for an android application/streaming service. The plan is that it is going to use x265enc as the encoder on the server and I have issues with constructing the pipeline for it. The pipeline currently looks like this: appsrc ! videoconvert ! x265enc ! mpegtsmux ! filesink ! appsrc has the settings: stream-type=0, is-live=true, do-timestamp=true, format=GST_FORMAT_TIME, caps: video/x-raw, format=RGB, width=640, height=420, framerate=30. x265enc has settings: speed-preset=1, bitrate=2000, tune=4 It does work perfectly encoding with H264 but I have issues when trying to implement H265. With these settings I don't get any data at all in the saved file. When I try with I420 as the format I do get a video but it is heavily distorted/green lines and is zoomed in. <<a href="http://gstreamer-devel.966125.n4.nabble.com/file/t379423/MR-leo.png" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/file/t379423/MR-leo.png</a>> We have tried running this pipeline in the terminal with the videotestsrc as the source instead of appsrc. If we do that it works so we figure that the issue has something to do with the appsrc. Any help is much appreciated! -- Sent from: <a href="http://gstreamer-devel.966125.n4.nabble.com/" rel="noreferrer" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/</a> ------------------------------ Message: 5 Date: Wed, 29 Apr 2020 14:29:45 -0500 From: William Johnston <<a href="mailto:wgj@cast.uark.edu" target="_blank">wgj@cast.uark.edu</a>> To: <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> Subject: Re: Streaming audio and video RTP Message-ID: <<a href="mailto:928e3bf0-4f0f-9df7-c5e3-ee46dc56858a@cast.uark.edu" target="_blank">928e3bf0-4f0f-9df7-c5e3-ee46dc56858a@cast.uark.edu</a>> Content-Type: text/plain; charset="utf-8"; Format="flowed" Careless of me, I linked it wrong. I linked the input of rtpbin to the input of udpsink. I'll try again: gst-launch-1.0 -e \ ??? ??? rtpbin name=rb ? ? ? ? uridecodebin uri="file:///home/fedora/starwars.mov" \ ? ? ? ? ! qtdemux name=demux ?demux.audio_0 \ ? ? ? ? ! queue \ ? ? ? ? ! audioconvert \ ? ? ? ? ! opusenc bandwidth=superwideband bitrate-type=vbr \ ? ? ? ? ! rtpopuspay ?\ ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5052 \ ? ? ? ? demux.video_0 \ ? ? ? ? ! queue \ ? ? ? ? ! videorate ! video/x-raw, framerate=30000/1001 \ ? ? ? ? ! videoconvert \ ? ? ? ? ! x264enc tune=zerolatency speed-preset=1 dct8x8=true quantizer=17 pass=qual \ ? ? ? ? ! rtph264pay \ ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ ? ? ? ? ! rb.send_rtp_sink_0 \ ??? ??? rb ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5054 \ On 4/28/2020 6:32 PM, Patrick Cusack wrote: > Ok. Good to know. Unfortunately, that doesn?t work. I get the following: > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > DtsGetHWFeatures: Create File Failed > DtsGetHWFeatures: Create File Failed > Running DIL (3.22.0) Version > DtsDeviceOpen: Opening HW in mode 0 > DtsDeviceOpen: Create File Failed > Redistribute latency... > WARNING: from element > /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0: Delayed linking > failed. > Additional debug info: > ./grammar.y(506): gst_parse_no_more_pads (): > /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0: > failed delayed linking some pad of GstURIDecodeBin named uridecodebin0 > to some pad of GstQTDemux named demux > Redistribute latency? > > I checked the stats on my server and don?t see any audio or video > packets coming in. The goal is to stream a file (eventually a video > input like Decklink) to a server that receives rtp. > > I can send audio or video separately and I don?t have issues. > > Patrick > >> On Apr 28, 2020, at 11:49 AM, William Johnston <<a href="mailto:wgj@cast.uark.edu" target="_blank">wgj@cast.uark.edu</a> >> <mailto:<a href="mailto:wgj@cast.uark.edu" target="_blank">wgj@cast.uark.edu</a>>> wrote: >> >> You can only specify ports on element names. Try this: >> >> gst-launch-1.0 -e \ >> ? ? ? ? uridecodebin uri="file:///home/fedora/starwars.mov" \ >> ? ? ? ? ! qtdemux name=demux ?demux.audio_0 \ >> ? ? ? ? ! queue \ >> ? ? ? ? ! audioconvert \ >> ? ? ? ? ! opusenc bandwidth=superwideband bitrate-type=vbr \ >> ? ? ? ? ! rtpopuspay ?\ >> ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ >> ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> >> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5052 \ >> ? ? ? ? demux.video_0 \ >> ? ? ? ? ! queue \ >> ? ? ? ? ! videorate ! video/x-raw, framerate=30000/1001 \ >> ? ? ? ? ! videoconvert \ >> ? ? ? ? ! x264enc tune=zerolatency speed-preset=1 dct8x8=true >> quantizer=17 pass=qual \ >> ? ? ? ? ! rtph264pay \ >> ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ >> ? ? ? ? ! rtpbin name=rb rb.send_rtp_sink_0 \ >> ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> >> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5054 \ >> >> >> On 4/28/2020 12:42 PM, Patrick Cusack wrote: >>> I have a endpoint that expects audio and video over ports 5052 and >>> 5054 respectively. I am using the following script to send audio and >>> video. I am getting a 'WARNING: erroneous pipeline: syntax error? >>> when I run the command. >>> Also, does using simple rtp payloads into a udp sink bypass RTCP >>> feedback, ie if my server is NACKing on account of dropped packets, >>> does this hinder retransmission of rtp packets? >>> >>> gst-launch-1.0 -e \ >>> ? ? ? ? uridecodebin uri="file:///home/fedora/starwars.mov" \ >>> ? ? ? ? ! qtdemux name=demux ?demux.audio_0 \ >>> ? ? ? ? ! queue \ >>> ? ? ? ? ! audioconvert \ >>> ? ? ? ? ! opusenc bandwidth=superwideband bitrate-type=vbr \ >>> ? ? ? ? ! rtpopuspay ?\ >>> ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ >>> ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> >>> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5052 \ >>> ? ? ? ? demux.video_0 \ >>> ? ? ? ? ! queue \ >>> ? ? ? ? ! videorate ! video/x-raw, framerate=30000/1001 \ >>> ? ? ? ? ! videoconvert \ >>> ? ? ? ? ! x264enc tune=zerolatency speed-preset=1 dct8x8=true >>> quantizer=17 pass=qual \ >>> ? ? ? ? ! rtph264pay \ >>> ? ? ? ? ! rtprtxqueue max-size-time=2000 max-size-packets=0 \ >>> ? ? ? ? ! rtpbin.send_rtp_sink_0 \ >>> ? ? ? ? ! udpsink host=<a href="http://www.playbacktc.com" rel="noreferrer" target="_blank">www.playbacktc.com</a> >>> <<a href="http://www.playbacktc.com/" rel="noreferrer" target="_blank">http://www.playbacktc.com/</a>> port=5054 \ >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> >>> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> >> _______________________________________________ >> gstreamer-devel mailing list >> <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> >> <mailto:<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a>> >> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> > > > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> > <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> -------------- next part -------------- An HTML attachment was scrubbed... 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