1. 24 Oct, 2008 3 commits
  2. 21 Oct, 2008 2 commits
  3. 16 Oct, 2008 3 commits
  4. 15 Oct, 2008 3 commits
  5. 10 Oct, 2008 1 commit
  6. 09 Oct, 2008 1 commit
  7. 08 Oct, 2008 5 commits
  8. 07 Oct, 2008 1 commit
  9. 03 Oct, 2008 4 commits
  10. 01 Oct, 2008 1 commit
    • Michael Smith's avatar
      configure.ac: Fix libs for linking directsound. · e2dbf108
      Michael Smith authored
      Original commit message from CVS:
      * configure.ac:
      Fix libs for linking directsound.
      * sys/directsound/gstdirectsoundsink.c:
      Fix buffer sizing to prevent racing the ringbuffer at startup.
      Add volume property.
      e2dbf108
  11. 27 Sep, 2008 1 commit
  12. 26 Sep, 2008 2 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit... · b17599a2
      Wim Taymans authored
      gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit signals a new talk spurt.
      
      Original commit message from CVS:
      * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init),
      (gst_rtp_amr_depay_process):
      Mark DISCONT on output buffers when the marker bit signals a new talk
      spurt.
      * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
      Set the marker bit for buffers with a DISCONT flag to signal a talk
      spurt.
      b17599a2
    • Wim Taymans's avatar
      gst/rtp/: Added MP4A-LATM payloader to match the depayloader. · c77bfaac
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
      (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
      (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
      (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
      (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
      (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
      * gst/rtp/gstrtpmp4apay.h:
      Added MP4A-LATM payloader to match the depayloader.
      c77bfaac
  13. 25 Sep, 2008 4 commits
  14. 23 Sep, 2008 2 commits
  15. 17 Sep, 2008 2 commits
    • Edward Hervey's avatar
      gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of... · 53a576bb
      Edward Hervey authored
      gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of zero... which basically results in qtdemux ...
      
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
      (gst_qtdemux_chain):
      Some 'broken' files out there have atom lengths of zero...
      which basically results in qtdemux consuming that atom again and again
      until the *end of night* !
      Detect that and emits an adequate element error message.
      53a576bb
    • Jan Schmidt's avatar
      gst/: Fix build flags order. · a236a2df
      Jan Schmidt authored
      Original commit message from CVS:
      * gst/interleave/Makefile.am:
      * gst/matroska/Makefile.am:
      Fix build flags order.
      * tests/check/elements/audioamplify.c: (GST_START_TEST):
      * tests/check/elements/audiodynamic.c: (GST_START_TEST):
      * tests/check/elements/audioinvert.c: (GST_START_TEST):
      * tests/check/elements/audiopanorama.c: (GST_START_TEST):
      Format fixes.
      * tests/check/elements/multifile.c:
      Pull in unistd.h
      a236a2df
  16. 15 Sep, 2008 2 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpmp4gdepay.*: Handle interleaved streams by reordering AU in a queue. · 1c6a371d
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
      (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
      (gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
      (gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
      (gst_rtp_mp4g_depay_change_state):
      * gst/rtp/gstrtpmp4gdepay.h:
      Handle interleaved streams by reordering AU in a queue.
      1c6a371d
    • Wim Taymans's avatar
      gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for... · fe9b4496
      Wim Taymans authored
      gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for the amount of bits we can use.
      
      Original commit message from CVS:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
      (gst_bs_parse_read), (gst_rtp_mp4g_depay_process):
      Change some of the ranges in the caps, mostly for the amount of bits we
      can use.
      Added a little bitstream parse and use it to parse the AU header fields.
      Check for malformed and wrongly sized packets better.
      Implement more header field parsing.
      Handle the size of fragmented packets correctly.
      fe9b4496
  17. 14 Sep, 2008 1 commit
  18. 11 Sep, 2008 1 commit
  19. 04 Sep, 2008 1 commit
    • Tim-Philipp Müller's avatar
      ext/flac/gstflacenc.c: Make sure the desired default values are actually set,... · 9a120212
      Tim-Philipp Müller authored
      ext/flac/gstflacenc.c: Make sure the desired default values are actually set, not only registered as defaults (actual...
      
      Original commit message from CVS:
      * ext/flac/gstflacenc.c: (gst_flac_enc_class_init):
      Make sure the desired default values are actually set, not only
      registered as defaults (actual problem is that the stereo-specific
      values are only updated if channels==2, which is not the case yet
      when the object is created, so the default values for the
      mid-side-stereo and loose-mid-side-stereo settings are never
      set in _update_quality()). Makes flacenc create smaller files by
      default (for stereo input), and fixes #550791.
      9a120212