1. 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
  2. 30 Jun, 2006 1 commit
  3. 29 Jun, 2006 1 commit
  4. 23 Jun, 2006 1 commit
    • Tim-Philipp Müller's avatar
      Use GST_DEBUG_CATEGORY_STATIC where possible (#342503). · 114a273f
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      * gst-libs/gst/rtp/gstbasertppayload.c:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      * gst/tcp/gstmultifdsink.c:
      * gst/tcp/gsttcpclientsink.c:
      * gst/tcp/gsttcpclientsrc.c:
      * gst/tcp/gsttcpserversink.c:
      * gst/tcp/gsttcpserversrc.c:
      * gst/videorate/gstvideorate.c:
      * gst/videotestsrc/gstvideotestsrc.c:
      * sys/v4l/gstv4ljpegsrc.c:
      * sys/v4l/gstv4lmjpegsink.c:
      * sys/v4l/gstv4lsrc.c:
      * tests/examples/seek/scrubby.c:
      * tests/examples/seek/seek.c:
      Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
      114a273f
  5. 22 Jun, 2006 3 commits
  6. 16 Jun, 2006 3 commits
    • Young-Ho Cha's avatar
      gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right... · f1392c14
      Young-Ho Cha authored
      gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
      
      Original commit message from CVS:
      Patch by: Young-Ho Cha <ganadist at chollian dot net>
      * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
      Parse extra data better, apparently it's right behind
      the normal strf header size. Fixes #343500.
      f1392c14
    • Tim-Philipp Müller's avatar
      Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file... · 5288476e
      Tim-Philipp Müller authored
      Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
      
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/cdda/gstcddabasesrc.h:
      Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
      out in the header file and shouldn't be listed in the docs.
      * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
      Fix it so that it doesn't crash in the debug statement.
      5288476e
    • Stefan Kost's avatar
      docs/libs/: add remaining symbols into correct setions · cade7911
      Stefan Kost authored
      Original commit message from CVS:
      * docs/libs/Makefile.am:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * docs/libs/gst-plugins-base-libs.types:
      add remaining symbols into correct setions
      * gst-libs/gst/audio/gstringbuffer.c:
      fix incomplete docs
      * gst-libs/gst/audio/gstringbuffer.h:
      comment out not yet implemented function
      * gst-libs/gst/floatcast/floatcast.h:
      * gst-libs/gst/netbuffer/gstnetbuffer.c:
      add short descriptions
      * gst-libs/gst/interfaces/propertyprobe.c:
      fix return value docs
      * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
      simplify debug logging
      * gst-libs/gst/riff/riff-read.h:
      sync function prototype and docs
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      remove left over symbol
      cade7911
  7. 14 Jun, 2006 1 commit
  8. 07 Jun, 2006 1 commit
    • Thomas Vander Stichele's avatar
      move last template doc snippets to source code and delete them · 51ca8fe3
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * docs/libs/tmpl/gstaudio.sgml:
      * docs/libs/tmpl/gstcolorbalance.sgml:
      * docs/libs/tmpl/gstmixer.sgml:
      * docs/libs/tmpl/gstringbuffer.sgml:
      * docs/libs/tmpl/gsttuner.sgml:
      * docs/libs/tmpl/gstxoverlay.sgml:
      * gst-libs/gst/audio/audio.c:
      * gst-libs/gst/audio/gstringbuffer.c:
      * gst-libs/gst/interfaces/colorbalance.c:
      * gst-libs/gst/interfaces/mixer.c:
      * gst-libs/gst/interfaces/tuner.c:
      * gst-libs/gst/interfaces/xoverlay.c:
      move last template doc snippets to source code and delete them
      51ca8fe3
  9. 03 Jun, 2006 1 commit
    • Jan Schmidt's avatar
      gst-libs/gst/audio/: Document better the fact that latency_time and... · 45e06fe7
      Jan Schmidt authored
      gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
      (gst_ring_buffer_samples_done):
      * gst-libs/gst/audio/gstringbuffer.h:
      Document better the fact that latency_time and buffer_time are values
      stored in microseconds, and not the usual GStreamer nanoseconds.
      Change the variables (compatibly) that store them from GstClockTime
      to guint64 to make it more clear that they're not storing clock times.
      Also, remove the bogus property description that says the user can
      specify -1 to get the default value, since that's never been the case.
      When computing the default segment size for the ring buffer, make it
      an integer number of samples.
      When the sub-class indicates a delay greater than the number of
      samples we've written return 0 from the audio sink get_time method.
      45e06fe7
  10. 01 Jun, 2006 1 commit
    • Stefan Kost's avatar
      Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass · 131fb86b
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixerelement.h:
      * ext/alsa/gstalsamixeroptions.h:
      * ext/alsa/gstalsamixertrack.h:
      * ext/gnomevfs/gstgnomevfssink.h:
      * ext/gnomevfs/gstgnomevfssrc.h:
      * ext/theora/gsttheoradec.h:
      * ext/theora/gsttheoraenc.h:
      * ext/theora/gsttheoraparse.h:
      * ext/vorbis/vorbisparse.h:
      * gst-libs/gst/audio/gstaudioclock.h:
      * gst-libs/gst/audio/gstaudiofilter.h:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      * gst/audioconvert/gstaudioconvert.h:
      * gst/audioresample/gstaudioresample.h:
      * gst/audiotestsrc/gstaudiotestsrc.h:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
      * gst/playback/gststreamselector.h:
      * gst/tcp/gstmultifdsink.h:
      * gst/tcp/gsttcpclientsink.h:
      * gst/tcp/gsttcpclientsrc.h:
      * gst/tcp/gsttcpserversink.h:
      * gst/tcp/gsttcpserversrc.h:
      * gst/videorate/gstvideorate.h:
      * gst/videoscale/gstvideoscale.h:
      * gst/videotestsrc/gstvideotestsrc.h:
      * gst/volume/gstvolume.h:
      * sys/v4l/gstv4ljpegsrc.h:
      * sys/v4l/gstv4lmjpegsink.h:
      * sys/v4l/gstv4lmjpegsrc.h:
      * sys/v4l/gstv4lsrc.h:
      * sys/ximage/ximagesink.h:
      * sys/xvimage/xvimagesink.h:
      * tests/old/testsuite/alsa/sinesrc.h:
      Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
      131fb86b
  11. 19 May, 2006 3 commits
  12. 18 May, 2006 1 commit
    • Philippe Kalaf's avatar
      gst-libs/gst/rtp/README: Some new documentation · 8675bc89
      Philippe Kalaf authored
      Original commit message from CVS:
      2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
      
      * gst-libs/gst/rtp/README:
      Some new documentation
      * gst-libs/gst/rtp/gstrtpbuffer.h:
      Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      New RTP audio base payloader class. Supports frame or sample based codecs.
      Not enabled in Makefile.am until approved.
      8675bc89
  13. 16 May, 2006 1 commit
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and... · 10d35563
      Tim-Philipp Müller authored
      gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      (gst_audio_check_channel_positions):
      It's okay to have caps with channels=1 and a channel position
      different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
      (deinterleavers might want to keep the position in the caps,
      so that they can be re-interleaved again properly later).
      Leave check for unexpected 2-channel layouts intact for now.
      10d35563
  14. 12 May, 2006 2 commits
    • Jan Schmidt's avatar
      Fix integer overflow problem with pixel-aspect-ratio calculations in... · 34db0838
      Jan Schmidt authored
      Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
      
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
      * gst-libs/gst/video/video.h:
      * gst/videoscale/Makefile.am:
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
      * tests/check/Makefile.am:
      * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
      (main):
      Fix integer overflow problem with pixel-aspect-ratio calculations
      in videoscale and xvimagesink (#341542)
      34db0838
    • Tim-Philipp Müller's avatar
      gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557). · 2f9b081b
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/tag/gstid3tag.c:
      Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
      2f9b081b
  15. 09 May, 2006 2 commits
    • Tim-Philipp Müller's avatar
      Const-ify GEnumValue and GFlagsValue arrays. Use · d8965c30
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
      (gst_text_overlay_halign_get_type),
      (gst_text_overlay_wrap_mode_get_type):
      * ext/theora/theoradec.c: (theora_handle_type_packet),
      (theora_handle_data_packet):
      * ext/theora/theoraenc.c: (gst_border_mode_get_type),
      (theora_enc_sink_setcaps), (theora_enc_chain):
      * gst-libs/gst/cdda/gstcddabasesrc.c:
      (gst_cdda_base_src_mode_get_type):
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audiostestsrc_wave_get_type):
      * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
      * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
      * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
      (gst_sync_method_get_type), (gst_unit_type_get_type),
      (gst_client_status_get_type):
      * gst/videoscale/gstvideoscale.c:
      (gst_video_scale_method_get_type):
      * gst/videotestsrc/gstvideotestsrc.c:
      (gst_video_test_src_pattern_get_type):
      * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
      (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
      (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
      (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
      (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
      (paint_setup_RGB565), (paint_setup_xRGB1555):
      Const-ify GEnumValue and GFlagsValue arrays. Use
      GST_ROUND_UP_* macros instead of home-made ones.
      d8965c30
    • Tim-Philipp Müller's avatar
      gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc. · 308f541b
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
      Add SEDG (Samsung MPEG-4) fourcc.
      308f541b
  16. 08 May, 2006 1 commit
  17. 02 May, 2006 1 commit
  18. 29 Apr, 2006 2 commits
  19. 28 Apr, 2006 4 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-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more... · 102b79e4
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      patch to make timestamp checking more tollerant to rounding
      errors given that real discontinuities are to be marked on
      buffers. Fixes some asf files and #338778.
      Also avoid some crashers when we receive an event in the
      NULL state.
      102b79e4
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing. · 04754176
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
      (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
      (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
      (gst_ring_buffer_pause), (gst_ring_buffer_stop),
      (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
      (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
      (gst_ring_buffer_commit), (gst_ring_buffer_read),
      (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
      (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
      Check arguments passed to public functions instead of
      crashing.
      04754176
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work. · c068425b
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
      (gst_base_audio_src_get_time), (gst_base_audio_src_create):
      GstBaseAudioSrc must be live or it does not work.
      * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
      Don't set live to TRUE as this is the default in the parentclass.
      c068425b
  20. 22 Apr, 2006 1 commit
  21. 13 Apr, 2006 2 commits
  22. 11 Apr, 2006 1 commit
    • Antoine Tremblay's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize,... · 5c7a0470
      Antoine Tremblay authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
      
      Original commit message from CVS:
      Patch by: Antoine Tremblay  <hexa00 at gmail dot com>
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
      Fix some memory leaks: on finalize, free buffers left in the queue
      before destroying the queue; in _push(), unref rtp_buf even if
      the process vfunc returned a NULL buffer as output buffer (#337548);
      demote some recuring debug messages to LOG level.
      5c7a0470
  23. 10 Apr, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not... · 35058f78
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_event):
      Starting the ringbuffer when we did not acquire it can cause
      a deadlock, is pointless and causes nasty things for
      subclasses.
      Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
      35058f78
  24. 08 Apr, 2006 2 commits
    • Stefan Kost's avatar
      Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) · 0afac375
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixeroptions.c:
      (gst_alsa_mixer_options_class_init):
      * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
      * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
      * gst-libs/gst/audio/gstaudiofilter.c:
      (gst_audio_filter_class_init):
      * gst-libs/gst/audio/gstaudiosink.c:
      (gst_audioringbuffer_class_init):
      * gst-libs/gst/audio/gstaudiosrc.c:
      (gst_audioringbuffer_class_init):
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
      * gst-libs/gst/interfaces/colorbalancechannel.c:
      (gst_color_balance_channel_class_init):
      * gst-libs/gst/interfaces/mixeroptions.c:
      (gst_mixer_options_class_init):
      * gst-libs/gst/interfaces/mixertrack.c:
      (gst_mixer_track_class_init):
      * gst-libs/gst/interfaces/tunerchannel.c:
      (gst_tuner_channel_class_init):
      * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
      * gst-libs/gst/netbuffer/gstnetbuffer.c:
      (gst_netbuffer_class_init):
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_class_init):
      * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
      * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
      * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_class_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
      * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
      * sys/v4l/gstv4lcolorbalance.c:
      (gst_v4l_color_balance_channel_class_init):
      * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
      * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
      * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
      * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
      (gst_v4l_tuner_norm_class_init):
      * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
      * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
      Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
      0afac375
    • Stefan Kost's avatar
      Fix broken GObject macros · 1a2642a1
      Stefan Kost authored
      Original commit message from CVS:
      * ext/pango/gsttextrender.h:
      * gst-libs/gst/audio/gstaudiosink.h:
      * gst-libs/gst/audio/gstaudiosrc.h:
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      * gst-libs/gst/audio/gstringbuffer.h:
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      * gst-libs/gst/rtp/gstbasertppayload.h:
      * gst-libs/gst/video/gstvideofilter.h:
      * gst-libs/gst/video/gstvideosink.h:
      * gst/playback/gstplaybasebin.h:
      * gst/tcp/gstmultifdsink.h:
      * sys/v4l/gstv4lelement.h:
      Fix broken GObject macros
      1a2642a1
  25. 24 Mar, 2006 1 commit
    • Stefan Kost's avatar
      Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top · 2d826700
      Stefan Kost 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:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
      (gst_gnome_vfs_sink_class_init):
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
      * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
      * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
      (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
      * ext/pango/gsttextoverlay.c:
      * ext/pango/gsttextrender.c:
      * ext/theora/theoradec.c:
      * ext/theora/theoraenc.c:
      * ext/vorbis/vorbisdec.c:
      * ext/vorbis/vorbisenc.c:
      * gst-libs/gst/audio/gstaudiofilter.c:
      (gst_audio_filter_base_init):
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      (gst_audio_filter_template_base_init):
      * gst/adder/gstadder.c: (gst_adder_get_type):
      * gst/adder/gstadder.h:
      * gst/audioconvert/gstaudioconvert.c:
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
      (gst_audio_test_src_create):
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstplaybin.c:
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_base_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
      * gst/volume/gstvolume.c:
      * sys/v4l/gstv4lmjpegsink.c:
      * sys/v4l/gstv4lmjpegsrc.c:
      * tests/check/libs/cddabasesrc.c:
      * tests/old/examples/gob/gst-identity2.gob:
      Add docs for adder, use GST_ELEMENT_DETAILS macro,
      define GstElementDetails at the top
      2d826700
  26. 23 Mar, 2006 1 commit