1. 22 Aug, 2007 1 commit
  2. 21 Aug, 2007 3 commits
  3. 17 Aug, 2007 2 commits
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the... · 846ddaa5
      Sebastian Dröge authored
      gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/audio.h:
      Use gst_util_uint64_scale() instead of doing the math
      with double for GST_FRAMES_TO_CLOCK_TIME() and
      GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
      prevents rounding errors. Fixes #467667.
      846ddaa5
    • Wim Taymans's avatar
      gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups. · 01d9553d
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
      (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
      * gst-libs/gst/rtsp/gstrtspconnection.h:
      Small cleanups.
      On shutdown, don't read the control socket yet.
      Set timeout value correctly in all cases.
      Add function to check if the server accepts reads or writes.
      API: gst_rtsp_connection_poll()
      * gst-libs/gst/rtsp/gstrtspdefs.h:
      Fix compilation with -pedantic.
      Add enum for _poll.
      01d9553d
  4. 16 Aug, 2007 2 commits
  5. 15 Aug, 2007 1 commit
  6. 13 Aug, 2007 3 commits
  7. 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
  8. 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
  9. 10 Aug, 2007 3 commits
  10. 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
  11. 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
  12. 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
  13. 06 Aug, 2007 1 commit
  14. 03 Aug, 2007 5 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
  15. 27 Jul, 2007 5 commits
  16. 26 Jul, 2007 4 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