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  1. Sep 23, 2019
  2. Sep 08, 2019
  3. Aug 06, 2019
  4. May 02, 2019
  5. Apr 18, 2019
  6. Apr 15, 2019
  7. Apr 11, 2019
    • Göran Jönsson's avatar
      rtsp_server: Free thread pool before clean transport cache · 3cfe8863
      Göran Jönsson authored
      If not waiting for free thread pool before clean transport caches, there
      can be a crash if a thread is executing in transport list loop in
      function send_tcp_message.
      
      Also add a check if priv->send_pool in on_message_sent to avoid that a
      new thread is pushed during wait of free thread pool. This is possible
      since when waiting for free thread pool mutex have to be unlocked.
      3cfe8863
  8. Apr 10, 2019
  9. Mar 27, 2019
  10. Mar 23, 2019
    • Tim-Philipp Müller's avatar
      g-i: pass --quiet to g-ir-scanner · 0becf0b6
      Tim-Philipp Müller authored
      This suppresses the annoying 'g-ir-scanner: link: cc ..' output
      that we get even if everything works just fine.
      
      We still get g-ir-scanner warnings and compiler warnings if
      we pass this option.
      0becf0b6
    • Tim-Philipp Müller's avatar
      g-i: silence 'nested extern' compiler warnings when building scanner binary · 6f434615
      Tim-Philipp Müller authored
      We need a nested extern in our init section for the scanner binary
      so we can call gst_init to make sure GStreamer types are initialised
      (they are not all lazy init via get_type functions, but some are in
      exported variables). There doesn't seem to be any other mechanism to
      achieve this, so just remove that warning, it's not important at all.
      6f434615
  11. Mar 21, 2019
  12. Mar 20, 2019
    • Göran Jönsson's avatar
      rtsp-media: Handle set state when preparing. · 1fd49d36
      Göran Jönsson authored
      Handle the situation when  a call to gst_rtsp_media_set_state is done
      when media status is preparing.
      
      Also add unit test for this scenario.
      
      The unit test simulate on a media level when two clients share a (live)
      media.
      Both clients have done SETUP and got responses. Now client 1 is doing
      play and client 2 is just closing the connection.
      
      Then without patch there are a problem when
      client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
      And client2 is doing closing connection we can end up in a call
      to gst_rtsp_media_set_state when
      priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
      shut down media is jumped over .
      
      With this patch and this scenario we wait until
      priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
      execute after that and now we will execute the logic for
      shut down media.
      1fd49d36
  13. Mar 04, 2019
  14. Feb 26, 2019
  15. Feb 19, 2019
  16. Feb 02, 2019
  17. Jan 30, 2019
  18. Jan 29, 2019
  19. Jan 25, 2019
    • Lars Wireen's avatar
      rtsp-media: Fix race codition in finish_unprepare · ae32203c
      Lars Wireen authored and Sebastian Dröge's avatar Sebastian Dröge committed
      The previous fix for race condition around finish_unprepare where the
      function could be called twice assumed that the status wouldn't change
      during execution of the function. This assumption is incorrect as the
      state may change, for example if an error message arrives from the
      pipeline bus.
      
      Instead a flag keeping track on whether the finish_unprepare function
      is currently executing is introduced and checked.
      
      Fixes #59
      ae32203c
  20. Jan 17, 2019
  21. Dec 06, 2018
  22. Dec 05, 2018
  23. Nov 20, 2018
  24. Nov 19, 2018
  25. Nov 17, 2018
  26. Nov 14, 2018
    • Linus Svensson's avatar
      rtsp-stream: Use seqnum-offset for rtpinfo · 18538592
      Linus Svensson authored and Sebastian Dröge's avatar Sebastian Dröge committed
      The sequence number in the rtpinfo is supposed to be the first RTP
      sequence number. The "seqnum" property on a payloader is supposed to be
      the number from the last processed RTP packet. The sequence number for
      payloaders that inherit gstrtpbasepayload will not be correct in case of
      buffer lists. In order to fix the seqnum property on the payloaders
      gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
      "seqnum-offset" from the "stats" property contains the value of the
      very first RTP packet in a stream. The server will, however, try to look
      at the last simple in the sink element and only use properties on the
      payloader in case there no sink elements yet, and by looking at the last
      sample of the sink gives the server full control of which RTP packet it
      looks at. If the payloader does not have the "stats" property, "seqnum"
      is still used since "seqnum-offset" is only present in as part of
      "stats" and this is still an issue not solved with this patch.
      
      Needed for gst-plugins-base!17
      18538592
    • Linus Svensson's avatar
      rtsp-stream: Plug memory leak · 1c4d3b36
      Linus Svensson authored and Sebastian Dröge's avatar Sebastian Dröge committed
      Attaching a GSource to a context will increase the refcount. The idle
      source will never be free'd since the initial reference is never
      dropped.
      1c4d3b36
  27. Nov 12, 2018
    • Jordan Petridіs's avatar
      Add Gitlab CI configuration · 2065298d
      Jordan Petridіs authored
      This commit adds a .gitlab-ci.yml file, which uses a feature
      to fetch the config from a centralized repository. The intent is
      to have all the gstreamer modules use the same configuration.
      
      The configuration is currently hosted at the gst-ci repository
      under the gitlab/ci_template.yml path.
      
      Part of gstreamer-project#29
      2065298d
  28. Nov 05, 2018
  29. Nov 01, 2018
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