1. 21 Aug, 2019 1 commit
  2. 17 Jul, 2019 1 commit
  3. 04 Jun, 2019 2 commits
  4. 01 Jun, 2019 1 commit
  5. 29 May, 2019 1 commit
  6. 25 May, 2019 1 commit
  7. 21 May, 2019 1 commit
  8. 17 May, 2019 1 commit
  9. 16 May, 2019 1 commit
  10. 13 May, 2019 3 commits
  11. 30 Apr, 2019 1 commit
  12. 19 Apr, 2019 1 commit
  13. 18 Apr, 2019 1 commit
  14. 10 Apr, 2019 2 commits
  15. 07 Apr, 2019 1 commit
  16. 05 Mar, 2019 1 commit
    • Tim-Philipp Müller's avatar
      audiodecoder: add _finish_subframe() method · 8d112201
      Tim-Philipp Müller authored
      This allows us to output audio samples without discarding
      any input frames, which is useful for some formats/codecs
      (e.g. the MonkeysAudio decoder implementation in ffmpeg
      which will might return e.g. 16 output buffers for an
      input buffer for certain files).
      In the past decoder implementations just concatenated
      the returned audio buffers until a full frame had been
      decoded, but that's no longer possible to do efficiently
      when the decoder returns audio samples in non-interleaved
      Allowing subframes to be output before the entire input
      frame is decoded can also be useful to decrease startup
  17. 04 Mar, 2019 1 commit
  18. 28 Feb, 2019 1 commit
  19. 26 Feb, 2019 1 commit
  20. 29 Jan, 2019 2 commits
  21. 17 Jan, 2019 1 commit
  22. 06 Jan, 2019 1 commit
  23. 28 Dec, 2018 3 commits
  24. 19 Dec, 2018 1 commit
  25. 17 Dec, 2018 1 commit
    • Mathieu Duponchelle's avatar
      audio-converter: add API to determine passthrough mode · 1edb2c42
      Mathieu Duponchelle authored
      audioconvert's passthrough status can no longer be determined
      strictly from input / output caps equality, as a mix-matrix can
      now be specified.
      We now call gst_base_transform_set_passthrough dynamically, based
      on the return from the new gst_audio_converter_is_passthrough()
      API, which takes the mix matrix into account.
  26. 15 Dec, 2018 1 commit
  27. 13 Dec, 2018 1 commit
    • Justin Kim's avatar
      rtcpbuffer: add support XR packet parsing · 5303e2c3
      Justin Kim authored
      According to RFC3611, the extended report blocks in XR packet can
      have variable length. To visit each block, the iterator should look
      into block header. Once XR type is extracted, users can parse the
      detailed information by given functions.
      Loss/Duplicate RLE
      The Loss RLE and the Duplicate RLE have same format so
      they can share parsers. For unit test, randomly generated
      pseudo packet is used.
      Packet Receipt Times
      The packet receipt times report block has a list of receipt
      times which are in [begin_seq, end_seq).
      Receiver Reference Time paser for XR packet
      The receiver reference time has ntptime which is 64 bit type.
      The DLRR report block consists of sub-blocks which has ssrc, last RR,
      and delay since last RR. The number of sub-blocks should be calculated
      from block length.
      Statistics Summary
      The Statistics Summary report block provides fixed length
      VoIP Metrics
      VoIP Metrics consists of several metrics even though they are in
      a report block. Data retrieving functions are added per metrics.
  28. 21 Nov, 2018 1 commit
    • Tomasz Andrzejak's avatar
      audiodecoder: add API for setting caps on the source pad · e0268c02
      Tomasz Andrzejak authored
      This patch adds API in the audio decoder base class for setting the arbitrary
      caps on the source pad.  Previously only caps converted from audio info were
      possible.  This is particularly useful when subclass wants to set caps features
      for audio decoder producing metadata.
  29. 12 Nov, 2018 1 commit
  30. 28 Oct, 2018 2 commits
  31. 10 Oct, 2018 1 commit
    • Stian Selnes's avatar
      rtpbasepayload: rtpbasedepayload: Add source-info property · f766b85b
      Stian Selnes authored
      Add a source-info property that will read/write meta to the buffers
      about RTP source information. The GstRTPSourceMeta can be used to
      transport information about the origin of a buffer, e.g. the sources
      that is included in a mixed audio buffer.
      A new function gst_rtp_base_payload_allocate_output_buffer() is added
      for payloaders to use to allocate the output RTP buffer with the correct
      number of CSRCs according to the meta and fill it.
      RTPSourceMeta does not make sense on RTP buffers since the information
      is in the RTP header. So the payloader will strip the meta from the
      output buffer.
  32. 03 Oct, 2018 1 commit