1. 24 Jan, 2021 1 commit
  2. 15 Jan, 2021 1 commit
    • Tobias Ronge's avatar
      rtspsrc: Do not wait for response while flushing · 706d9137
      Tobias Ronge authored
      Due to the may_cancel flag in GstRTSPConnection, receiving might not get
      cancelled when supposed to. In this case, gst_rtsp_src_receive_response
      will have to wait until timeout instead but if busy receiving RTP
      data, this timeout will never occur.
      
      With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
      if flushing is set to TRUE instead of continuing to receive.
      
      Part-of: <gstreamer/gst-plugins-good!831>
      706d9137
  3. 14 Jan, 2021 2 commits
  4. 11 Jan, 2021 1 commit
  5. 04 Jan, 2021 4 commits
  6. 30 Dec, 2020 1 commit
  7. 21 Dec, 2020 1 commit
  8. 16 Dec, 2020 1 commit
  9. 14 Dec, 2020 1 commit
  10. 12 Dec, 2020 3 commits
  11. 11 Dec, 2020 2 commits
  12. 08 Dec, 2020 2 commits
  13. 04 Dec, 2020 1 commit
  14. 01 Dec, 2020 1 commit
  15. 26 Nov, 2020 1 commit
  16. 24 Nov, 2020 1 commit
    • Nirbheek Chauhan's avatar
      deinterlace: Enable x86 assembly with nasm on MSVC · 552da856
      Nirbheek Chauhan authored
      We need to remove x86inc.asm from the list of compiled assembly files
      because it is not supposed to be compiled separately. It is directly
      included by yadif.asm, and it exports no symbols.
      
      The object file was getting ignored on all platforms except on msvc
      where it was causing a linker hang when building with debugging
      enabled because the object file had no debug symbols (or similar).
      We've seen this before in FFmpeg too, which uses nasm:
      gstreamer/meson-ports/ffmpeg!46
      
      Part-of: <!825>
      552da856
  17. 20 Nov, 2020 1 commit
  18. 19 Nov, 2020 1 commit
  19. 16 Nov, 2020 1 commit
  20. 13 Nov, 2020 3 commits
    • Håvard Graff's avatar
      rtpsession: never send on a non-internal source · 79748dab
      Håvard Graff authored
      This will end up as a "received" packet, due to the code in
      source_push_rtp, which will think this is a packet being received.
      
      Instead drop the packet and hope that either:
      1. Something upstream responds to the GstRTPCollision event and changes
         SSRC used for sending.
      2. That the application responds to the "on-ssrc-collision" signal, and
         forces the sender (payloader) to change its SSRC.
      3. That the BYE sent to the existing user of this SSRC will respond to
         the BYE, and that we timeout this source, so we can continue sending
         using the chosen SSRC.
      
      The test reproduces a scenario where we previously would have sent
      on a non-internal source.
      
      Part-of: <!817>
      79748dab
    • Håvard Graff's avatar
      rtpsource: rewrite timeout-check to avoid underflow · 97ced292
      Håvard Graff authored
      If current_time is < collision_timeout, we get an uint64 underflow, and
      the check will trigger prematurely.
      
      Part-of: <gstreamer/gst-plugins-good!817>
      97ced292
    • Vivia Nikolaidou's avatar
      aacparse: Fix caps change handling · 5a2f9d51
      Vivia Nikolaidou authored
      In baseparse we set the fixed caps flag on all src pads, therefore the
      source pad caps query in get_allowed_caps will return the current caps.
      Current caps won't necessarily intersect with the new caps (e.g. sample
      rate change). Replace get_allowed_caps with peer_query_caps.
      
      Part-of: <gstreamer/gst-plugins-good!816>
      5a2f9d51
  21. 12 Nov, 2020 2 commits
  22. 11 Nov, 2020 2 commits
  23. 10 Nov, 2020 2 commits
  24. 04 Nov, 2020 1 commit
  25. 03 Nov, 2020 3 commits