1. 16 Aug, 2007 1 commit
  2. 15 Aug, 2007 1 commit
  3. 14 Aug, 2007 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/app/gstappsink.c: Don't use new API. · 1ec11dbc
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
      Don't use new API.
      1ec11dbc
    • Wim Taymans's avatar
      gst-libs/gst/app/gstappsink.*: Make love to appsink. · dd72f88a
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
      (gst_app_sink_class_init), (gst_app_sink_dispose),
      (gst_app_sink_flush_unlocked), (gst_app_sink_start),
      (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
      (gst_app_sink_render), (gst_app_sink_get_caps),
      (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
      (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
      * gst-libs/gst/app/gstappsink.h:
      Make love to appsink.
      Make it support pulling of the preroll buffer.
      Add docs and debug statements.
      Fix some races wrt to EOS handling and stopping.
      Implement getcaps.
      Implement FLUSHING.
      API: gst_app_sink_pull_preroll()
      dd72f88a
  4. 13 Aug, 2007 3 commits
  5. 12 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that... · 3b7071a1
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_set_outcaps):
      * gst-libs/gst/rtp/gstbasertppayload.h:
      Improve caps negotiation so that downstream elements can confiure
      certain RTP properties by fixing them on the caps. See #465146.
      Add docs.
      3b7071a1
  6. 11 Aug, 2007 1 commit
    • Tim-Philipp Müller's avatar
      Mark as deprecated some macros which were presumably meant to be private API... · 2d5d5ac8
      Tim-Philipp Müller authored
      Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
      
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      Mark as deprecated some macros which were presumably meant to be
      private API and accidentally exposed in the public header file.
      Also actually _init() lock (only works at the moment because the
      struct is zeroed out when created and the initial values in the
      mutex struct are zeroes too). (#459585)
      2d5d5ac8
  7. 10 Aug, 2007 3 commits
  8. 09 Aug, 2007 4 commits
    • Michael Smith's avatar
      gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity,... · 9f9e76bc
      Michael Smith authored
      gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
      
      Original commit message from CVS:
      * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
      If we have a large (> 1 second) discontinuity, push a series of
      smaller buffers rather than a single very large buffer. Avoids
      unreasonably large single buffer allocations when encountering a
      large gap.
      * tests/check/elements/audiorate.c: (GST_START_TEST),
      (audiorate_suite):
      Add a test for this.
      9f9e76bc
    • Josep Torra Valles's avatar
      gst/playback/gstplaybasebin.c: Fixes: #465015 · 9730f452
      Josep Torra Valles authored
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c: (group_commit),
      (queue_remove_probe), (queue_threshold_reached):
      Patch by: Josep Torra Valles <josep@fluendo.com>
      Fixes: #465015
      Make sure we remove the check_queues buffer probe from the
      correct queue to avoid racily going back to "buffering 99%" when
      buffering is actually complete.
      Also, fix the spelling of Josep's surname in the ChangeLog.
      9730f452
    • Stefan Kost's avatar
      ext/ogg/gstoggmux.c: Do not leak oggmux instance. · 87d96c65
      Stefan Kost authored
      Original commit message from CVS:
      * ext/ogg/gstoggmux.c:
      Do not leak oggmux instance.
      * ext/vorbis/vorbisenc.c:
      Also log values.
      87d96c65
    • Thomas Vander Stichele's avatar
      po/: Updated translations. · f6be63b9
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * po/hu.po:
      * po/it.po:
      * po/nl.po:
      * po/uk.po:
      * po/vi.po:
      Updated translations.
      f6be63b9
  9. 08 Aug, 2007 2 commits
    • Yang Hong's avatar
      ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979 · afd8b931
      Yang Hong authored
      Original commit message from CVS:
      patch by: Yang Hong <hongyang@redflag-linux.com>
      * ext/pango/gsttextoverlay.c:
      * ext/pango/gsttextoverlay.h:
      Add 'silent' property to GstTimeOverlay. Fixes #462979
      afd8b931
    • Josep Torre Valles's avatar
      Add connection-speed property. Fixes #464690. · 382b7102
      Josep Torre Valles authored
      Original commit message from CVS:
      Patch by: Josep Torre Valles <josep@fluendo.com>
      * docs/plugins/gst-plugins-base-plugins.args:
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
      (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
      (gst_uri_decode_bin_get_property), (gen_source_element):
      Add connection-speed property. Fixes #464690.
      382b7102
  10. 07 Aug, 2007 2 commits
    • Damien Lespiau's avatar
      Fix compilation on windows. Fixes #464320. · 9b8c8371
      Damien Lespiau authored
      Original commit message from CVS:
      Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
      * configure.ac:
      * gst-libs/gst/rtsp/Makefile.am:
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      (gst_rtsp_connection_connect):
      Fix compilation on windows. Fixes #464320.
      9b8c8371
    • Josep Torre Valles's avatar
      gst/playback/: Move connection-speed property from playbin to playbasebin so... · 5e5aa7b4
      Josep Torre Valles authored
      gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
      
      Original commit message from CVS:
      Patch by: Josep Torre Valles <josep@fluendo.com>
      * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
      (gst_play_base_bin_init), (queue_threshold_reached),
      (gen_source_element), (setup_substreams),
      (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
      (gst_play_base_bin_get_streaminfo_value_array):
      * gst/playback/gstplaybasebin.h:
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
      (gst_play_bin_set_property), (gst_play_bin_get_property),
      (gst_play_bin_handle_redirect_message):
      Move connection-speed property from playbin to playbasebin so that we
      can also configure it in source elements that have the connection-speed
      property. Fixes #464028.
      Add some debug info here and there.
      5e5aa7b4
  11. 06 Aug, 2007 1 commit
  12. 03 Aug, 2007 6 commits
    • Sebastian Dröge's avatar
      gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to... · 6f397125
      Sebastian Dröge authored
      gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
      
      Original commit message from CVS:
      * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
      (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
      (gst_audio_test_src_init_sine_table),
      (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
      * gst/audiotestsrc/gstaudiotestsrc.h:
      Add float/double and int32 support to audiotestsrc. Fixes #460422.
      Also set the default volume to the default value specified in the
      GParamSpec.
      6f397125
    • Jens Granseuer's avatar
      gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of... · ef33f2fd
      Jens Granseuer authored
      gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
      
      Original commit message from CVS:
      Patch by: Jens Granseuer <jensgr at gmx dot net>
      * gst/audioconvert/gstaudioquantize.c:
      Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
      ef33f2fd
    • Wim Taymans's avatar
      gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport. · 607fa48a
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
      Add rdt manager for rdt transport.
      Fix parsing of RDT transport.
      607fa48a
    • Jan Schmidt's avatar
      configure.ac: Back to CVS · 2f6e0e7b
      Jan Schmidt authored
      Original commit message from CVS:
      * configure.ac:
      Back to CVS
      2f6e0e7b
    • Jan Schmidt's avatar
      Release 0.10.14 · 221ae4eb
      Jan Schmidt authored
      Original commit message from CVS:
      Release 0.10.14
      221ae4eb
    • Jan Schmidt's avatar
      Update .po files · 42771c4f
      Jan Schmidt authored
      Original commit message from CVS:
      Update .po files
      42771c4f
  13. 27 Jul, 2007 5 commits
  14. 26 Jul, 2007 6 commits
    • Jan Schmidt's avatar
      gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as... · aa14635c
      Jan Schmidt authored
      gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
      
      Original commit message from CVS:
      * gst-libs/gst/interfaces/Makefile.am:
      * gst-libs/gst/interfaces/interfaces-marshal.list:
      * gst-libs/gst/interfaces/rtspextension.c:
      * gst-libs/gst/interfaces/rtspextension.h:
      * gst-libs/gst/rtsp/Makefile.am:
      * gst-libs/gst/rtsp/gstrtsp.h:
      * gst-libs/gst/rtsp/gstrtspextension.c:
      (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
      (gst_rtsp_extension_detect_server),
      (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
      (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
      (gst_rtsp_extension_configure_stream),
      (gst_rtsp_extension_get_transports),
      (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
      * gst-libs/gst/rtsp/gstrtspextension.h:
      * gst-libs/gst/rtsp/rtsp-marshal.list:
      Move the rtspextension.h interface into gstrtspextension.h
      as part of libgstrtsp instead of libgstinterfaces, because it's
      only for use within plugins, not applications.
      Add stuff to do the enum & marshal generation needed in libgstrtsp now.
      Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
      signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
      is abstract.
      aa14635c
    • Wim Taymans's avatar
      gst-libs/gst/interfaces/: Fix marshaller for the send signal. · 6d1a34ef
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/interfaces/Makefile.am:
      * gst-libs/gst/interfaces/interfaces-marshal.list:
      * gst-libs/gst/interfaces/rtspextension.c:
      (gst_rtsp_extension_iface_init),
      (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
      * gst-libs/gst/interfaces/rtspextension.h:
      Fix marshaller for the send signal.
      Add URL to stream selection interface method.
      6d1a34ef
    • Jan Schmidt's avatar
      gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside. · 50a3a239
      Jan Schmidt authored
      Original commit message from CVS:
      * gst-libs/gst/riff/Makefile.am:
      Pull in our dependencies from -base before those from outside.
      50a3a239
    • Wim Taymans's avatar
      API: gst_rtsp_base64_decode_ip() · 2c35823b
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
      * gst-libs/gst/rtsp/gstrtspbase64.h:
      API: gst_rtsp_base64_decode_ip()
      Added function to decode Base64 in-place.
      2c35823b
    • Jan Schmidt's avatar
      tests/check/libs/.cvsignore: Ignore the mixer test binary. · 58afe32d
      Jan Schmidt authored
      Original commit message from CVS:
      * tests/check/libs/.cvsignore:
      Ignore the mixer test binary.
      58afe32d
    • Jan Schmidt's avatar
      ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots. · b947924e
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
      Gratuitous comment change to trigger a rebuild on the buildbots.
      b947924e
  15. 25 Jul, 2007 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can. · 8db50d49
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
      (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
      (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
      (gst_sdp_media_get_format), (gst_sdp_media_get_information),
      (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
      (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
      (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
      (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
      (gst_sdp_media_get_attribute_val):
      * gst-libs/gst/sdp/gstsdpmessage.h:
      Constify args where we can.
      8db50d49
    • Wim Taymans's avatar
      gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here. · 256d005e
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/interfaces/Makefile.am:
      * gst-libs/gst/interfaces/rtspextension.c:
      (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
      (gst_rtsp_extension_detect_server),
      (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
      (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
      (gst_rtsp_extension_configure_stream),
      (gst_rtsp_extension_get_transports),
      (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
      * gst-libs/gst/interfaces/rtspextension.h:
      Move interface for RTSP extensions from -good to here.
      Added helper methods to invoke interface methods.
      256d005e