1. 10 Dec, 2008 2 commits
  2. 09 Dec, 2008 2 commits
  3. 08 Dec, 2008 4 commits
    • Wim Taymans's avatar
      gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the... · cf0efcbf
      Wim Taymans authored
      gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
      
      Original commit message from CVS:
      * gst/volume/gstvolume.c: (gst_volume_class_init),
      (volume_before_transform), (volume_transform_ip):
      Use new basetransform vmethod to reconfigure the dynamic properties and
      any pending volume/mute changes. Fixes #563508.
      cf0efcbf
    • Sebastian Dröge's avatar
      configure.ac: First check for "theoraenc theoradec" and if that failed check... · 9912a227
      Sebastian Dröge authored
      configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
      
      Original commit message from CVS:
      * configure.ac:
      First check for "theoraenc theoradec" and if that failed check
      for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
      deprecate the latter. Also linking on Windows fails with just "theora"
      and the version check would fail for the release candidates.
      Fixes bug #563718.
      9912a227
    • Stefan Kost's avatar
      gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2. · 16e2bccc
      Stefan Kost authored
      Original commit message from CVS:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstdecodebin2.c:
      Add basic docs to decodebin and link to decodebin from decodebin2.
      16e2bccc
    • Olivier Crete's avatar
      gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174. · 3c9df39c
      Olivier Crete authored
      Original commit message from CVS:
      Patch by: Olivier Crete  <tester at tester ca>
      * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
      * gst-libs/gst/rtp/gstrtcpbuffer.h:
      Implement gst_rtcp_packet_remove(). Fixes #563174.
      * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
      Add unit test for some RTCP functions.
      3c9df39c
  4. 04 Dec, 2008 2 commits
  5. 03 Dec, 2008 1 commit
  6. 01 Dec, 2008 1 commit
    • 이문형's avatar
      gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we... · 933186aa
      이문형 authored
      gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
      
      Original commit message from CVS:
      Patch by: 이문형 <iwings at gmail dot com>
      * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
      Don't forget to release the lock again if we bail out because some
      pad is flushing or we've reached EOS, otherwise things will lock up
      next time _push_buffer() is called (#562802).
      933186aa
  7. 29 Nov, 2008 1 commit
    • Sebastian Dröge's avatar
      Require gettext 0.17 because older versions don't mix with libtool 2.2. At... · 97b2cac6
      Sebastian Dröge authored
      Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
      
      Original commit message from CVS:
      Patch by: Cygwin Ports maintainer
      <yselkowitz at users dot sourceforge dot net>
      * autogen.sh:
      * configure.ac:
      Require gettext 0.17 because older versions don't mix with libtool
      2.2. At build time an older gettext version will still work.
      Fixes bug #556091.
      97b2cac6
  8. 28 Nov, 2008 4 commits
  9. 27 Nov, 2008 5 commits
    • Sebastian Dröge's avatar
      Remove audioresample files. · 7afac6e2
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/audioresample/Makefile.am:
      * gst/audioresample/buffer.c:
      * gst/audioresample/buffer.h:
      * gst/audioresample/debug.c:
      * gst/audioresample/debug.h:
      * gst/audioresample/functable.c:
      * gst/audioresample/functable.h:
      * gst/audioresample/gstaudioresample.c:
      * gst/audioresample/gstaudioresample.h:
      * gst/audioresample/resample.c:
      * gst/audioresample/resample.h:
      * gst/audioresample/resample_chunk.c:
      * gst/audioresample/resample_functable.c:
      * gst/audioresample/resample_ref.c:
      * tests/check/elements/audioresample.c:
      Remove audioresample files.
      7afac6e2
    • Sebastian Dröge's avatar
      docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change. · 32ceb10b
      Sebastian Dröge authored
      Original commit message from CVS:
      * docs/plugins/inspect/plugin-audioresample.xml:
      Regenerated for library filename change.
      32ceb10b
    • 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
      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
  10. 25 Nov, 2008 6 commits
  11. 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
      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
  12. 22 Nov, 2008 3 commits
  13. 21 Nov, 2008 3 commits
    • Jonathan Rosser's avatar
      gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report... · 60262606
      Jonathan Rosser authored
      gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978).  Try 'video...
      
      Original commit message from CVS:
      Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
      * gst/videotestsrc/gstvideotestsrc.c:
      * gst/videotestsrc/gstvideotestsrc.h:
      * gst/videotestsrc/videotestsrc.c:
      * gst/videotestsrc/videotestsrc.h:
      Add a zone plate pattern generator based on BBC R&D Report
      1978/23 (yeah *that* 1978).  Try 'videotestsrc pattern=zone-plate
      kx2=20 ky2=20 kt=1'.
      60262606
    • Sebastian Dröge's avatar
      gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps... · d36adc54
      Sebastian Dröge authored
      gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
      
      Original commit message from CVS:
      * gst/speexresample/gstspeexresample.c:
      (gst_speex_resample_class_init), (gst_speex_resample_set_property),
      (gst_speex_resample_get_property):
      Add a "filter-length" property that maps to the quality values
      for compatibilty with audioresample.
      d36adc54
    • Michael Smith's avatar
      gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile. · 5830b42d
      Michael Smith authored
      Original commit message from CVS:
      * gst/playback/gstdecodebin2.c:
      Fix random fat-fingering making this not compile.
      5830b42d