1. 02 Feb, 2022 2 commits
  2. 28 Jan, 2022 1 commit
  3. 18 Jan, 2022 3 commits
  4. 17 Jan, 2022 4 commits
    • Rafał Dzięgiel's avatar
      assrender: Support RFC8081 mime types · 7d19b935
      Rafał Dzięgiel authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      Old "application/*" are now as per RFC8081 deprecated in favor of
      new "font/*" mime types. Some new encoders are already using the
      updated mime types. We need to also add them to the support list
      in order for assrender to correctly identify them as fonts.
      
      Part-of: <!2566>
      7d19b935
    • Rafał Dzięgiel's avatar
      assrender: Add "application/vnd.ms-opentype" mimetype detection · f7d7c575
      Rafał Dzięgiel authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      The "application/vnd.ms-opentype" mimetype is commonly used in many fonts attached in the matroska videos.
      Assrender should treat it as compatible without the need of parsing the file extension.
      
      Part-of: <!2566>
      f7d7c575
    • Rafał Dzięgiel's avatar
      assrender: Handle ".ttc" attachment extension · 8a7c698c
      Rafał Dzięgiel authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      TTC stands for "TrueType Collection" file. We can pass it
      into libass as any other attachment. Add it to the supported
      extensions list, so the fonts it contains will be used.
      
      Part-of: <!2566>
      8a7c698c
    • Thibault Saunier's avatar
      pitch: Specify layout as required for negotiation · b877c72a
      Thibault Saunier authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      There are cases where it might negotiate 'non-interleaved' while it
      is wrong.
      
      ```
      gst-launch-1.0 audiotestsrc !  "audio/x-raw, format=(string)F32LE, layout=(string)non-interleaved" ! audioconvert ! audioresample ! pitch tempo=1.2 ! audioconvert ! "audio/x-raw,format=S16LE" ! fakesink
      
      Setting pipeline to PAUSED ...
      Pipeline is PREROLLING ...
      (gst-launch-1.0:3029628): GStreamer-Audio-CRITICAL **: 11:42:22.477: gst_audio_buffer_map: assertion '(!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta && info->layout == meta->info.layout)' failed
      ERROR: from element /GstPipeline:pipeline0/GstAudioConvert:audioconvert1: The stream is in the wrong format.
      Additional debug info:
      ../subprojects/gst-plugins-base/gst/audioconvert/gstaudioconvert.c(876): gst_audio_convert_transform (): /GstPipeline:pipeline0/GstAudioConvert:audioconvert1:
      failed to map input buffer
      ERROR: pipeline doesn't want to preroll.
      ERROR: from element /GstPipeline:pipeline0/GstAudioTe...
      b877c72a
  5. 05 Nov, 2021 1 commit
  6. 31 Oct, 2021 3 commits
  7. 29 Oct, 2021 2 commits
    • Vivia Nikolaidou's avatar
      mpegtspacketizer: memcmp potentially seen_before data · 5c000cb0
      Vivia Nikolaidou authored and GStreamer Marge Bot's avatar GStreamer Marge Bot committed
      Theoretically the version number is incremented every time there's a new
      section, but in a world of streaming we can't easily make that
      assumption.
      
      An example of a broken use case is when we're cat-ing two mpeg-ts files
      together, which is equivalent of capturing a DVB stream while switching
      channels. A set-top box would know that we switched the channels and
      reset the demuxer, but in practice this might not happen.
      
      Part-of: <!2555>
      5c000cb0
    • Stéphane Cerveau's avatar
      zxing: update to support version 1.1.1 · 54029f8f
      Stéphane Cerveau authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      Support new API in 1.1.1
      Update the supported input video format.
      Update tests to use parse_launch
      
      Part-of: <!2554>
      54029f8f
  8. 21 Oct, 2021 1 commit
  9. 08 Sep, 2021 3 commits
  10. 06 Sep, 2021 3 commits
  11. 02 Sep, 2021 1 commit
  12. 30 Aug, 2021 1 commit
  13. 25 Aug, 2021 2 commits
    • Tim-Philipp Müller's avatar
      openh264enc: fix broken header AU emission by base class · 36a09411
      Tim-Philipp Müller authored
      This encoder advertises alignment=au as output format, which means
      each output frame should contain a full decodable access unit.
      
      The video encoder base class is not aware of our output alignment
      and will output spurious buffers with just the SPS/PPS inside when
      we call gst_video_encoder_set_headers(), which is broken because
      each buffer is supposed to contain a full decodable access unit
      in our case.
      
      Just don't tell the base class about our headers, they will be
      sent at the beginning of each IDR frame anyway.
      
      Part-of: <!2478>
      36a09411
    • Tim-Philipp Müller's avatar
      openh264enc: fix header buffer leak · a91c5670
      Tim-Philipp Müller authored
      Part-of: <!2478>
      a91c5670
  14. 24 Aug, 2021 1 commit
    • Tim-Philipp Müller's avatar
      openh264enc: fix broken sps/pps header generation · 5ff895d3
      Tim-Philipp Müller authored
      This was putting a truncated SPS into the initial header instead
      of the PPS because it was always reading from the beginning of the
      bitstream buffer (pBsBuf) and not from the offset where the current
      NAL is at in the bitstream buffer (psBsBuf + nal_offset).
      
      This was broken in commit 17113695.
      
      Fixes #1576
      
      Part-of: <!2478>
      5ff895d3
  15. 23 Aug, 2021 1 commit
  16. 20 Aug, 2021 1 commit
  17. 18 Aug, 2021 1 commit
  18. 09 Aug, 2021 1 commit
  19. 29 Jul, 2021 1 commit
  20. 16 Jul, 2021 1 commit
  21. 09 Jul, 2021 1 commit
  22. 08 Jul, 2021 5 commits