1. 27 Nov, 2008 3 commits
    • Sebastian Dröge's avatar
      Rename the moved speexresample to audioresample, integrate into the build... · 153406ee
      Sebastian Dröge authored
      Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
      
      Original commit message from CVS:
      * configure.ac:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      * docs/plugins/gst-plugins-base-plugins.args:
      * docs/plugins/gst-plugins-base-plugins.hierarchy:
      * docs/plugins/gst-plugins-base-plugins.interfaces:
      * docs/plugins/gst-plugins-base-plugins.prerequisites:
      * docs/plugins/inspect/plugin-adder.xml:
      * docs/plugins/inspect/plugin-alsa.xml:
      * docs/plugins/inspect/plugin-audioconvert.xml:
      * docs/plugins/inspect/plugin-audiorate.xml:
      * docs/plugins/inspect/plugin-audioresample.xml:
      * docs/plugins/inspect/plugin-audiotestsrc.xml:
      * docs/plugins/inspect/plugin-cdparanoia.xml:
      * docs/plugins/inspect/plugin-decodebin.xml:
      * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
      * docs/plugins/inspect/plugin-gdp.xml:
      * docs/plugins/inspect/plugin-gio.xml:
      * docs/plugins/inspect/plugin-gnomevfs.xml:
      * docs/plugins/inspect/plugin-libvisual.xml:
      * docs/plugins/inspect/plugin-ogg.xml:
      * docs/plugins/inspect/plugin-pango.xml:
      * docs/plugins/inspect/plugin-playback.xml:
      * docs/plugins/inspect/plugin-queue2.xml:
      * docs/plugins/inspect/plugin-subparse.xml:
      * docs/plugins/inspect/plugin-tcp.xml:
      * docs/plugins/inspect/plugin-theora.xml:
      * docs/plugins/inspect/plugin-typefindfunctions.xml:
      * docs/plugins/inspect/plugin-uridecodebin.xml:
      * docs/plugins/inspect/plugin-video4linux.xml:
      * docs/plugins/inspect/plugin-videorate.xml:
      * docs/plugins/inspect/plugin-videoscale.xml:
      * docs/plugins/inspect/plugin-videotestsrc.xml:
      * docs/plugins/inspect/plugin-volume.xml:
      * docs/plugins/inspect/plugin-vorbis.xml:
      * docs/plugins/inspect/plugin-ximagesink.xml:
      * docs/plugins/inspect/plugin-xvimagesink.xml:
      * gst/speexresample/gstspeexresample.c: (plugin_init):
      * gst/speexresample/Makefile.am:
      * tests/check/Makefile.am:
      * tests/check/elements/speexresample.c: (setup_speexresample),
      (GST_START_TEST), (test_pipeline):
      Rename the moved speexresample to audioresample, integrate into the
      build system and remove the old audioresample from the build system.
      Fixes bug #558124, #385061, #346218, #116051.
      153406ee
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12... · af354dbe
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
      Avoid nasty int overflows after about 12 hours and 25 minutes when these
      code paths are triggered.
      A free beer to Håvard Graff for finding this!
      af354dbe
    • 이문형's avatar
      gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't... · d80a5c9d
      이문형 authored and Wim Taymans's avatar Wim Taymans committed
      gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
      
      Original commit message from CVS:
      Patch by: 이문형 <iwings at gmail dot com>
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      (gst_rtsp_connection_connect):
      A successful gst_poll_wait() doesn't always mean successful connect() on
      Windows.  We should check errors by calling gst_poll_fd_has_error().
      See #561924.
      d80a5c9d
  2. 25 Nov, 2008 6 commits
  3. 24 Nov, 2008 6 commits
    • Michael Smith's avatar
      gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to... · aec03a45
      Michael Smith authored
      gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
      
      Original commit message from CVS:
      * gst/playback/gstplaybin2.c:
      Add notification of current stream. Add ability to configure buffer
      sizes.
      * gst/playback/gsturidecodebin.c:
      Add ability to configure buffer sizes for streaming mode.
      Bug #561734.
      aec03a45
    • Stefan Kost's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove... · a8264f66
      Stefan Kost authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      Time is already in running_time. Remove base_time handling. Fixes
      audiosinks not draining and thus chopping some audio in the end.
      a8264f66
    • David Schleef's avatar
      ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture. · 3d894ebe
      David Schleef authored
      Original commit message from CVS:
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggmux.h:
      If we're muxing a dirac stream, flush the page after every picture.
      3d894ebe
    • Stefan Kost's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for... · 7f937c99
      Stefan Kost authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      Add one log message to check for audio_drained. Sync one log message
      with the condition. Send EOS after draining audio in pull mode.
      7f937c99
    • Sebastian Dröge's avatar
      ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation... · 79bb2ffe
      Sebastian Dröge authored
      ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
      
      Original commit message from CVS:
      * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
      Use gst_buffer_try_new_and_alloc() and fail properly if the
      allocation failed. This prevents abort() if downstream elements
      request an insane amount of memory.
      79bb2ffe
    • Jon Trowbridge's avatar
      gst/volume/gstvolume.*: Cleanup volume, define and use default values. · 0bdeaae5
      Jon Trowbridge authored and Wim Taymans's avatar Wim Taymans committed
      Original commit message from CVS:
      * gst/volume/gstvolume.c: (volume_choose_func),
      (volume_update_volume), (gst_volume_set_volume),
      (gst_volume_get_volume), (gst_volume_set_mute),
      (gst_volume_class_init), (gst_volume_init),
      (volume_process_double), (volume_process_float),
      (volume_process_int32), (volume_process_int32_clamp),
      (volume_process_int24), (volume_process_int24_clamp),
      (volume_process_int16), (volume_process_int16_clamp),
      (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
      (volume_transform_ip), (volume_set_property),
      (volume_get_property):
      * gst/volume/gstvolume.h:
      Cleanup volume, define and use default values.
      Recalculate new volume and mute setup before processing. Fixes #561789.
      * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
      Add controller unit test. Patch by: Jonathan Matthew
      Fix bogus test that messed with basetransform's internal state.
      0bdeaae5
  4. 22 Nov, 2008 3 commits
  5. 21 Nov, 2008 3 commits
  6. 20 Nov, 2008 4 commits
  7. 19 Nov, 2008 1 commit
    • David Schleef's avatar
      gst/videotestsrc/: Add "colorspec" property, specifying whether to generate... · b97e582c
      David Schleef authored
      gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video.  This only affect...
      
      Original commit message from CVS:
      * gst/videotestsrc/gstvideotestsrc.c:
      * gst/videotestsrc/gstvideotestsrc.h:
      * gst/videotestsrc/videotestsrc.c:
      * gst/videotestsrc/videotestsrc.h:
      Add "colorspec" property, specifying whether to generate BT.601
      or BT.709 video.  This only affects YCbCr values, not RGB, since
      if you're generating a 709 test pattern, presumably you want
      709 RGB primaries, not 601.  Also add "smpte75" pattern, which
      uses 75% colors instead of 100%, since this is often more useful
      for testing (and also follows the SMPTE EG-1 guideline).
      b97e582c
  8. 18 Nov, 2008 1 commit
  9. 14 Nov, 2008 3 commits
  10. 13 Nov, 2008 4 commits
    • Mark Nauwelaerts's avatar
      gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types. · 23f10c54
      Mark Nauwelaerts authored
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
      (plugin_init):
      Improve typefinding of ISO JPEG2000 mime types.
      23f10c54
    • Wim Taymans's avatar
      sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts. · e02bde49
      Wim Taymans authored
      Original commit message from CVS:
      * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
      (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
      (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
      * sys/xvimage/xvimagesink.h:
      Avoid typechecking when we do trivial casts.
      Move error handling out of the main program flow.
      Sneak in the display-region caps property, not completely correct yet.
      Cache the width/height in buffer_alloc instead of parsing it from the
      caps all the time.
      e02bde49
    • Wim Taymans's avatar
      gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we... · 2773fe8f
      Wim Taymans authored
      gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
      
      Original commit message from CVS:
      * gst/playback/gstplaybin2.c: (deactivate_group):
      don't try to unlink the selector sinkpad when we don't have it yet. This
      can happen if an error occured before the group was complete.
      2773fe8f
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as... · 9c32e1f1
      Wim Taymans authored
      gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
      (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
      (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
      (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
      (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
      (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
      (gst_rtp_buffer_get_extension_data),
      (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
      (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
      (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
      (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
      (gst_rtp_buffer_get_payload_type),
      (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
      (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
      (gst_rtp_buffer_set_timestamp),
      (gst_rtp_buffer_get_payload_subbuffer),
      (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
      Avoid expensive type checks we already did as part of the
      _validate() function that should be called first.
      9c32e1f1
  11. 11 Nov, 2008 2 commits
  12. 10 Nov, 2008 4 commits