1. 14 Apr, 2009 1 commit
  2. 25 Mar, 2009 1 commit
  3. 19 Dec, 2008 1 commit
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type()... · 4ed1f5d6
      Sebastian Dröge authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      Make GstAudioSrcSlaveMethod get_type() function non-static
      as it's public now.
      * win32/common/libgstaudio.def:
      * win32/common/libgstnetbuffer.def:
      Add some missing functions to the list of exported symbols.
      4ed1f5d6
  4. 13 Dec, 2008 1 commit
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to... · 04d9ff9a
      Sebastian Dröge authored
      gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_slave_method_get_type),
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_slave_method_get_type),
      (gst_base_audio_src_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
      public API. This is needed for the C++ bindings to be able
      to use this base classes. Fixes bug #564200, #564206.
      04d9ff9a
  5. 27 Nov, 2008 1 commit
    • 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
  6. 08 Oct, 2008 2 commits
  7. 13 Aug, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also... · 510a5bef
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      When not slaved to another clock also subtract the base_time from our
      internal clock time to get the running time.
      510a5bef
  8. 07 Aug, 2008 1 commit
    • Frederic Crozat's avatar
      Make sure gettext returns translations in UTF-8 encoding rather than in the... · 89be2461
      Frederic Crozat authored
      Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
      
      Original commit message from CVS:
      Patch by: Frederic Crozat <fcrozat@mandriva.org>
      * ext/alsa/gstalsaplugin.c: (plugin_init):
      * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
      * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
      * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
      * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
      * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
      * gst/playback/gstdecodebin.c: (plugin_init):
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
      * gst/playback/gstplayback.c: (plugin_init):
      * gst/playback/gstqueue2.c: (plugin_init):
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
      * sys/v4l/gstv4l.c: (plugin_init):
      Make sure gettext returns translations in UTF-8 encoding rather
      than in the current locale encoding (#546822).
      89be2461
  9. 31 May, 2008 2 commits
    • Mark Nauwelaerts's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new... · 9fa61c52
      Mark Nauwelaerts authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      Add a gtk-doc chunk for the new properties to have a Since: indication.
      9fa61c52
    • Mark Nauwelaerts's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and... · c660bbd6
      Mark Nauwelaerts authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
      (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
      (gst_base_audio_src_change_state):
      Provide readable actual-buffer-time and actual-latency-time properties
      that reflect the configured ringbuffer values. Fixes #524724.
      c660bbd6
  10. 27 May, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the... · 35e4b75b
      Wim Taymans authored
      gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
      (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
      * gst-libs/gst/audio/gstaudioclock.h:
      Add method to inform the clock that the time starts from 0 again. We use
      this info to calculate a clock offset so that the time we report in
      internal_time is monotonically increasing, as required by the clock base
      class. Fixes #521761.
      API: GstAudioClock::gst_audio_clock_reset()
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_skew_slaving),
      (gst_base_audio_sink_change_state):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create), (gst_base_audio_src_change_state):
      Reset reported time when we (re)create the ringbuffer.
      35e4b75b
  11. 20 May, 2008 1 commit
    • Wim Taymans's avatar
      ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they... · d8dc371c
      Wim Taymans authored
      ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
      
      Original commit message from CVS:
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
      (gst_gnome_vfs_src_finalize),
      (gst_gnome_vfs_src_received_headers_callback),
      (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
      * ext/gnomevfs/gstgnomevfssrc.h:
      Set the ICY caps on the srcpad from where they get picked up by the base
      class now and set on the outgoing buffers.
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
      BaseSrc now sets the caps on outgoing buffers automatically.
      d8dc371c
  12. 28 Apr, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs. · 7916e386
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Clarify some docs.
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
      (gst_base_audio_src_class_init), (gst_base_audio_src_init),
      (gst_base_audio_src_set_slave_method),
      (gst_base_audio_src_get_slave_method),
      (gst_base_audio_src_set_property),
      (gst_base_audio_src_get_property), (gst_base_audio_src_create):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      Add property and methods for selecting the clock slave method in the
      source, like in the sink.
      We only implement "none" and "re-timestamp" for now.
      API: gst_base_audio_src_set_slave_method()
      API: gst_base_audio_src_get_slave_method()
      7916e386
  13. 06 Apr, 2008 1 commit
    • Tim-Philipp Müller's avatar
      gst/: Work around missing bits of thread-safety on older GLibs some more to... · 7a29d716
      Tim-Philipp Müller authored
      gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      * gst/playback/gstplayback.c: (plugin_init):
      * gst/volume/gstvolume.c: (plugin_init):
      Work around missing bits of thread-safety on older GLibs some
      more to avoid assertions when starting up multiple playbin
      objects concurrently (see #512382).
      7a29d716
  14. 22 Mar, 2008 1 commit
    • Sebastian Dröge's avatar
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings... · 49deb0c0
      Sebastian Dröge authored
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
      
      Original commit message from CVS:
      * configure.ac:
      * ext/alsa/gstalsamixerelement.c:
      (gst_alsa_mixer_element_class_init):
      * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
      * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
      * ext/cdparanoia/gstcdparanoiasrc.c:
      (gst_cd_paranoia_src_class_init):
      * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
      * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
      * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
      * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
      * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
      * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
      * ext/pango/gsttextrender.c: (gst_text_render_class_init):
      * ext/theora/theoradec.c: (gst_theora_dec_class_init):
      * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
      * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
      * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      (gst_audio_filter_template_class_init):
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      * gst-libs/gst/cdda/gstcddabasesrc.c:
      (gst_cdda_base_src_class_init):
      * gst-libs/gst/interfaces/mixertrack.c:
      (gst_mixer_track_class_init):
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_class_init):
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_class_init):
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_class_init):
      * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
      * gst/audioresample/gstaudioresample.c:
      (gst_audioresample_class_init):
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audio_test_src_class_init):
      * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
      * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
      (preroll_unlinked):
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
      * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
      * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
      * gst/playback/gstqueue2.c: (gst_queue_class_init):
      * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
      * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
      (gst_stream_selector_class_init):
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_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/videorate/gstvideorate.c: (gst_video_rate_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):
      * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
      * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
      * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
      * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
      * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
      static strings (i.e. all). This gives us less memory usage,
      fewer allocations and thus less memory defragmentation. Depend
      on core CVS for this. Fixes bug #523806.
      49deb0c0
  15. 10 Mar, 2008 1 commit
  16. 10 Jan, 2008 1 commit
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make... · 3feb4bc8
      Tim-Philipp Müller authored
      gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      Ref audio clock class from a thread-safe context to make sure
      we're not bit by GObjects lack of thread-safety here (#349410),
      however unlikely that may be in practice.
      3feb4bc8
  17. 17 Dec, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info. · 2ea251a3
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create), (gst_base_audio_src_change_state):
      Add debug info.
      When going from PLAYING to PAUSED, pause the ringbuffer before calling
      the parent state change function, just like the audiosink, because the
      parent waits for the element to finish its processing before completing
      the state change. This makes going to PAUSED a lot snappier.
      When going from READY to PAUSED, don't allow the ringbuffer to start
      yet.
      2ea251a3
  18. 21 Nov, 2007 1 commit
    • Wim Taymans's avatar
      Expose methods for some object properties so that subclasses can more easily configure them. · 157a65b1
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
      (gst_base_audio_sink_set_provide_clock),
      (gst_base_audio_sink_get_provide_clock),
      (gst_base_audio_sink_set_slave_method),
      (gst_base_audio_sink_get_slave_method),
      (gst_base_audio_sink_set_property),
      (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
      (gst_base_audio_sink_none_slaving),
      (gst_base_audio_sink_handle_slaving):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Expose methods for some object properties so that subclasses can more
      easily configure them.
      Added slave method none, that completely disables slaving to the
      internal clock.
      API: gst_base_audio_sink_set_provide_clock()
      API: gst_base_audio_sink_get_provide_clock()
      API: gst_base_audio_sink_set_slave_method()
      API: gst_base_audio_sink_get_slave_method()
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_set_provide_clock),
      (gst_base_audio_src_get_provide_clock),
      (gst_base_audio_src_set_property),
      (gst_base_audio_src_get_property), (gst_base_audio_src_create):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      Expose methods for some object properties so that subclasses can more
      easily configure them.
      API: gst_base_audio_src_set_provide_clock()
      API: gst_base_audio_src_get_provide_clock()
      157a65b1
  19. 08 Oct, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no... · c3dda05a
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      Also handle the case where there is no clock set on the audio source,
      like in the unit tests.
      c3dda05a
  20. 10 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal... · c9422524
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_init),
      (gst_base_audio_src_provide_clock),
      (gst_base_audio_src_set_property),
      (gst_base_audio_src_get_property), (gst_base_audio_src_create):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      Allow othe clocks than the internal clock to be used for the pipeline.
      Add property to disable clock provide.
      API: GstBaseAudioSrc::provide-clock
      c9422524
  21. 21 May, 2007 1 commit
    • Wim Taymans's avatar
      Small cleanups. · 9b188adc
      Wim Taymans authored
      Original commit message from CVS:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      (gst_cd_paranoia_src_read_sector):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      Small cleanups.
      * ext/theora/theoradec.c: (theora_dec_sink_event):
      Fix typo.
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_set_gst_timestamp):
      Add some FIXME
      * gst/playback/gstdecodebin.c: (queue_underrun_cb):
      And some debug info when a FIXME path is hit.
      9b188adc
  22. 25 Apr, 2007 1 commit
    • Tim-Philipp Müller's avatar
      When posting a warning message because samples were dropped, post something... · 9e873a3c
      Tim-Philipp Müller authored
      When posting a warning message because samples were dropped, post something more intelligible than he default error m...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
      (gst_base_audio_src_create):
      * po/POTFILES.in:
      When posting a warning message because samples were dropped, post
      something more intelligible than he default error message for clock
      errors which is just confusing in this context (#432984).
      9e873a3c
  23. 28 Feb, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudioclock.c: Fix clock name. · 3c94c06c
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
      (gst_audio_clock_new):
      Fix clock name.
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_init), (gst_base_audio_sink_query):
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
      (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
      (gst_base_audio_src_create):
      Improve latency query code.
      Use proper clock names.
      3c94c06c
  24. 15 Feb, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query. · a43d0f57
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
      (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
      (gst_base_audio_sink_async_play),
      (gst_base_audio_sink_change_state):
      Answer latency query.
      Use configured latency when syncing.
      Fix clock slaving.
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
      (gst_base_audio_src_query), (gst_base_audio_src_change_state):
      Fix possible memleak.
      Implement latency query.
      Small cleanups.
      a43d0f57
  25. 12 Jan, 2007 1 commit
    • Andy Wingo's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c · d853b238
      Andy Wingo authored
      Original commit message from CVS:
      2007-01-12  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstbaseaudiosink.c
      (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
      (gst_base_audio_sink_activate_pull): Remove the handwavey nego
      stuff, as the base class handles this now. Actually tell the ring
      buffer to start.
      (gst_base_audio_sink_callback): Cast the ring buffer correctly.
      How did this work before? Maybe I'm not as awesome a programmer as
      I think.
      
      * gst-libs/gst/audio/gstbaseaudiosrc.c
      (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
      of a pad function.
      d853b238
  26. 27 Sep, 2006 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING. · 1980f167
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      Add some more info in a WARNING.
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      Handle PAUSE in create function, use new -core addition to
      wait for playing. Fixes pausing and resuming capture from an
      audiosrc.
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
      (gst_ring_buffer_read):
      Constify some more.
      Caller supports interrupted reads now.
      1980f167
    • Wim Taymans's avatar
      Added docs for the audio libs. · 73677225
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/audio/gstaudioclock.c:
      * gst-libs/gst/audio/gstaudioclock.h:
      * gst-libs/gst/audio/gstaudiosink.c:
      * gst-libs/gst/audio/gstaudiosink.h:
      * gst-libs/gst/audio/gstaudiosrc.c:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      * gst-libs/gst/audio/gstringbuffer.h:
      Added docs for the audio libs.
      73677225
  27. 15 Sep, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes... · 65b1938b
      Wim Taymans authored
      gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
      (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
      (gst_base_audio_src_create), (gst_base_audio_src_change_state):
      Do the delay calculation in the source/sink base classes as this is
      specific for the capture/playback mode.
      Try to fixate a bit better, like round depth up to a multiple of 8
      bigger than width.
      Handle underruns correctly by marking DISCONT on buffers and adjusting
      timestamps to handle the gap.
      Set offset/offset_end correctly on buffers.
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
      (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
      (gst_ring_buffer_read):
      Remove resync and underrun recovery from the ringbuffer.
      Fix ringbuffer read code on under/overrun.
      65b1938b
  28. 12 Jul, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when... · a0354a5b
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_set_clock),
      (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
      Don't try to post an error message when setting the clock fails
      as this can happen when adding an element to a bin which will then
      deadlock. Fixes #347296.
      a0354a5b
  29. 06 Jul, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass) · fa5dacc9
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init),
      (gst_base_audio_sink_provide_clock):
      Use gobject_class instead of G_OBJECT_CLASS (klass)
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_init),
      (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
      (gst_base_audio_src_get_time),
      (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
      (gst_base_audio_src_create_ringbuffer):
      Fix latency and buffer-time constants and properties ala basesink.
      Implement pull based scheduling. Fixes #346527.
      Set default blocksize in GstBaseSrc to 0, we default to pushing out
      one segment.
      Refuse slaving to another clock instead of silently not working.
      Only provide a clock when we are actually able to do so.
      Various small cleanups and compiler hints.
      fa5dacc9
  30. 28 Apr, 2006 1 commit
  31. 23 Mar, 2006 2 commits
  32. 25 Jan, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging. · 2bc5ca17
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
      (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
      Improve debugging.
      Post error when caps cannot be parsed.
      Resync on discontinuity in the stream.
      Clip samples to segment boundaries.
      return WRONG_STATE sooner when we are flushing.
      
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
      (gst_base_audio_src_get_time), (gst_base_audio_src_create):
      Make audiosrc operate in TIME.
      Set TIMESTAMP and DURATION on buffers.
      2bc5ca17
  33. 20 Dec, 2005 2 commits
    • Thomas Vander Stichele's avatar
      stop making fun of older compilers · 01bc68f9
      Thomas Vander Stichele authored
      Original commit message from CVS:
      stop making fun of older compilers
      01bc68f9
    • Thomas Vander Stichele's avatar
      gst-libs/gst/audio/: update strings, values are in microseconds change the... · b4b2b62a
      Thomas Vander Stichele authored
      gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
      
      Original commit message from CVS:
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      update strings, values are in microseconds
      change the default sink buffer time to something that is smaller
      (to help software volume mixing have a slightly lower delay) but
      still be acceptable on Wim's laptop
      b4b2b62a
  34. 06 Dec, 2005 1 commit
  35. 21 Nov, 2005 1 commit
    • Jan Schmidt's avatar
      Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) · 1cc82e91
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/libvisual/visual.c: (get_buffer):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_fixate):
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_fixate_caps):
      * gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audiotestsrc_src_fixate):
      * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
      * gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
      * gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
      * gst/videotestsrc/gstvideotestsrc.c:
      (gst_videotestsrc_src_fixate):
      * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
      Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
      (#322027)
      1cc82e91