1. 28 Oct, 2006 1 commit
  2. 20 Aug, 2006 1 commit
    • Stefan Kost's avatar
      gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size · c2d7af84
      Stefan Kost authored
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_get_unit_size), (set_structure_widths):
      Lower debug, use g_assert in _get_unit_size
      * gst/audioresample/gstaudioresample.c:
      (audioresample_get_unit_size):
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      (gst_ffmpegcsp_get_unit_size):
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
      use g_assert in _get_unit_size
      c2d7af84
  3. 28 Jul, 2006 1 commit
    • Jan Schmidt's avatar
      gst/audioresample/gstaudioresample.c: Don't leak references to the incoming... · e828178e
      Jan Schmidt authored
      gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
      
      Original commit message from CVS:
      * gst/audioresample/gstaudioresample.c: (audioresample_stop),
      (audioresample_set_caps):
      Don't leak references to the incoming caps. Clean them up when
      stopping.
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
      (gst_video_scale_finalize):
      Don't leak our temporary pixel buffer.
      * tests/check/Makefile.am:
      * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
      (GST_START_TEST), (simple_launch_lines_suite):
      Fix leaks and re-enable the test for valgrind checking.
      e828178e
  4. 22 Jun, 2006 1 commit
    • Cody Russell's avatar
      gst/: Avoid unnecessary class cast check in class_init functions (#337747). · c10584ed
      Cody Russell authored
      Original commit message from CVS:
      Patch by: Cody Russell <bratsche at gnome org>
      * gst/audioresample/gstaudioresample.c:
      (gst_audioresample_class_init):
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_class_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
      * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
      * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
      * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
      * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
      * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
      * gst/videotestsrc/gstvideotestsrc.c:
      (gst_video_test_src_class_init):
      * gst/volume/gstvolume.c: (gst_volume_class_init):
      Avoid unnecessary class cast check in class_init
      functions (#337747).
      c10584ed
  5. 16 Jun, 2006 1 commit
    • Tim-Philipp Müller's avatar
      gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and... · 767ac56e
      Tim-Philipp Müller authored
      gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
      
      Original commit message from CVS:
      * gst/audioresample/gstaudioresample.c:
      (gst_audioresample_class_init), (gst_audioresample_init),
      (audioresample_start), (audioresample_stop),
      (gst_audioresample_set_property), (gst_audioresample_get_property):
      Implement GstBaseTransform::start and ::stop so that audioresample
      can clear its internal state properly and be reused insted of
      causing non-negotiated errors with playbin under some circumstances
      (#342789).
      * tests/check/elements/audioresample.c: (setup_audioresample),
      (cleanup_audioresample):
      Need to set element state here so that ::start and ::stop are
      called.
      767ac56e
  6. 28 Apr, 2006 2 commits
    • Stefan Kost's avatar
      make GstElementDetails const · e972defd
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixerelement.c:
      * ext/alsa/gstalsasrc.c:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      * ext/gnomevfs/gstgnomevfssink.c:
      * ext/gnomevfs/gstgnomevfssrc.c:
      * ext/ogg/gstoggdemux.c:
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggparse.c:
      * ext/ogg/gstogmparse.c:
      * ext/pango/gstclockoverlay.c:
      * ext/pango/gsttextoverlay.c:
      * ext/pango/gsttextrender.c:
      * ext/pango/gsttimeoverlay.c:
      * ext/theora/theoradec.c:
      * ext/theora/theoraenc.c:
      * ext/vorbis/vorbisdec.c:
      * ext/vorbis/vorbisenc.c:
      * gst-libs/gst/audio/gstaudiofilter.c:
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      * gst/audioconvert/gstaudioconvert.c:
      * gst/audiorate/gstaudiorate.c:
      * gst/audioresample/gstaudioresample.c:
      * gst/audiotestsrc/gstaudiotestsrc.c:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstplaybin.c:
      * gst/playback/gststreamselector.c:
      * gst/subparse/gstsubparse.c:
      * gst/tcp/gstmultifdsink.c:
      * gst/tcp/gsttcpclientsink.c:
      * gst/tcp/gsttcpclientsrc.c:
      * gst/tcp/gsttcpserversink.c:
      * gst/tcp/gsttcpserversrc.c:
      * gst/typefind/gsttypefindfunctions.c: (plugin_init):
      * gst/videorate/gstvideorate.c:
      * gst/videoscale/gstvideoscale.c:
      * gst/videotestsrc/gstvideotestsrc.c:
      * gst/volume/gstvolume.c:
      * sys/v4l/gstv4ljpegsrc.c:
      * sys/v4l/gstv4lmjpegsink.c:
      * sys/v4l/gstv4lmjpegsrc.c:
      * sys/v4l/gstv4lsrc.c:
      * sys/ximage/ximagesink.c:
      * sys/xvimage/xvimagesink.c:
      * tests/check/libs/cddabasesrc.c:
      make GstElementDetails const
      e972defd
    • Wim Taymans's avatar
      gst/audioresample/gstaudioresample.c: Add support for other formats... · 8cd920fc
      Wim Taymans authored
      gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
      
      Original commit message from CVS:
      * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
      (resample_set_state_from_caps):
      Add support for other formats audioresample can handle such as
      32 bits in and float and 64 bits float. Fixes #301759
      8cd920fc
  7. 02 Mar, 2006 1 commit
    • Wim Taymans's avatar
      docs/plugins/: Add audioresample to docs. · af09257f
      Wim Taymans authored
      Original commit message from CVS:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-docs.sgml:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      Add audioresample to docs.
      * gst/audioconvert/gstaudioconvert.c:
      Add revision date.
      * gst/audioresample/gstaudioresample.c:
      (gst_audioresample_base_init), (gst_audioresample_class_init),
      (gst_audioresample_init), (gst_audioresample_dispose),
      (audioresample_get_unit_size), (audioresample_transform_caps),
      (resample_set_state_from_caps), (audioresample_transform_size),
      (audioresample_set_caps), (audioresample_event),
      (audioresample_do_output), (audioresample_transform),
      (audioresample_pushthrough), (gst_audioresample_set_property),
      (gst_audioresample_get_property), (plugin_init):
      * gst/audioresample/gstaudioresample.h:
      Added docs.
      Small code cleanups.
      af09257f
  8. 15 Dec, 2005 1 commit
  9. 06 Dec, 2005 1 commit
  10. 02 Dec, 2005 1 commit
    • Wim Taymans's avatar
      gst/audioresample/: Fix audioresample, seek torture, new segments, reverse... · e3a77670
      Wim Taymans authored
      gst/audioresample/: Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
      
      Original commit message from CVS:
      * gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
      * gst/audioresample/buffer.h:
      * gst/audioresample/gstaudioresample.c:
      * gst/audioresample/gstaudioresample.h:
      * gst/audioresample/resample.c: (resample_input_flush),
      (resample_input_pushthrough), (resample_input_eos),
      (resample_get_output_size_for_input),
      (resample_get_input_size_for_output), (resample_get_output_size),
      (resample_get_output_data):
      * gst/audioresample/resample.h:
      * gst/audioresample/resample_ref.c: (resample_scale_ref):
      Fix audioresample, seek torture, new segments, reverse negotiation
      etc.. work fine.
      e3a77670
  11. 21 Nov, 2005 1 commit
    • Wim Taymans's avatar
      gst/: Segment update fix. · 0f2336cf
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
      (gst_base_audio_sink_provide_clock),
      (gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
      (gst_base_audio_sink_change_state):
      * gst/audioresample/gstaudioresample.c:
      Segment update fix.
      0f2336cf
  12. 16 Oct, 2005 1 commit
  13. 23 Sep, 2005 1 commit
  14. 09 Sep, 2005 1 commit
    • Jan Schmidt's avatar
      check/: Add extra tests for basetransform based components. · 0f4fa24d
      Jan Schmidt authored
      Original commit message from CVS:
      * check/Makefile.am:
      * check/pipelines/simple_launch_lines.c: (setup_pipeline),
      (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
      Add extra tests for basetransform based components.
      Comment out the test_element_negotiation test until we decide
      if it's testing correct behaviour.
      * ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
      (gst_visual_chain), (gst_visual_change_state):
      Slightly more correct but still bogus timestamping.
      Fix state change function.
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_class_init):
      * gst/audioresample/gstaudioresample.c:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      (gst_ffmpegcsp_class_init):
      * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
      (gst_videoscale_prepare_size), (gst_videoscale_set_caps),
      (gst_videoscale_prepare_image):
      * gst/volume/gstvolume.c: (gst_volume_class_init),
      (volume_transform_ip):
      Basetransform updates. Enable passthrough modes.
      * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
      (gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
      (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
      Negotiation fix that allows the window to return to the original
      size and renegotiate passthrough upstream. Extra debug output.
      0f4fa24d
  15. 28 Aug, 2005 1 commit
  16. 26 Aug, 2005 1 commit
  17. 25 Aug, 2005 3 commits
    • Thomas Vander Stichele's avatar
      check/: add a test for audioconvert · 6dff9c2c
      Thomas Vander Stichele authored
      Original commit message from CVS:
      
      * check/Makefile.am:
      * check/elements/audioconvert.c: (setup_audioconvert),
      (cleanup_audioconvert), (get_int_caps), (verify_convert),
      (GST_START_TEST), (audioconvert_suite), (main):
      add a test for audioconvert
      * gst/audioresample/gstaudioresample.c:
      * gst/audioresample/gstaudioresample.h:
      set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
      note that for buffers of 1/3 sec this means DURATION(c) is
      one nanosecond more than for a and b
      6dff9c2c
    • Thomas Vander Stichele's avatar
      add a check for audioresample · eae12502
      Thomas Vander Stichele authored
      Original commit message from CVS:
      add a check for audioresample
      eae12502
    • Thomas Vander Stichele's avatar
      gst/audioresample/: add room for extra overlap samples when asked to transform... · 7647f7fc
      Thomas Vander Stichele authored
      gst/audioresample/: add room for extra overlap samples when asked to transform size protect against possible mem corr...
      
      Original commit message from CVS:
      * gst/audioresample/debug.c:
      * gst/audioresample/gstaudioresample.c:
      add room for extra overlap samples when asked to transform size
      protect against possible mem corruption and check for discrepancies
      between written size and outbuffer's size so we can warn for
      potential problems
      * gst/audioresample/resample.c: (resample_init),
      (resample_get_output_size_for_input), (resample_get_output_size),
      (resample_set_n_channels), (resample_set_format):
      set debug level based on RESAMPLE_DEBUG env var
      make sure that get_output_size* returns a whole number of
      sample_size
      set sample_size each time either channel or format is set
      * gst/audioresample/resample_chunk.c: (resample_scale_chunk):
      * gst/audioresample/resample_functable.c:
      (resample_scale_functable):
      * gst/audioresample/resample_ref.c: (resample_scale_ref):
      remove r->sample_size, it's done in resample.c now
      add some debugging to the ref implementation
      make sure we only give back bytes that are wholes of the sample
      size
      7647f7fc
  18. 24 Aug, 2005 1 commit
  19. 23 Aug, 2005 1 commit
    • David Schleef's avatar
      gst/audioresample/Makefile.am: Leet audioresampling code · ae8f41b6
      David Schleef authored
      Original commit message from CVS:
      * gst/audioresample/Makefile.am: Leet audioresampling code
      * 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:
      ae8f41b6