mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-08 15:45:55 +00:00
Code Issues 1 Projects Releases Wiki Activity
6075 commits 44 branches 1505 tags 314 MiB
Commit graph

184 commits

Author SHA1 Message Date
Stefan Kost e972defd3e make GstElementDetails const 
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
 2006-04-28 19:46:37 +00:00
Wim Taymans f9356758c8 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float):
correctly clip float samples > 1.0. Fixes #338718
 2006-04-28 14:12:28 +00:00
j^ 08047f5cfe better/unified long descriptions 
Original commit message from CVS:
Patch by: j^ <j at bootlab dot org>
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.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/audioconvert/gstaudioconvert.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:
better/unified long descriptions
Fixes #336477
 2006-03-29 14:00:08 +00:00
Stefan Kost 2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top 
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
 2006-03-24 10:42:11 +00:00
Wim Taymans af09257fd0 docs/plugins/: Add audioresample to docs. 
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.
 2006-03-02 18:23:55 +00:00
Wim Taymans 8011ea8e2d Fix some more docs. 
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/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
* ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
(vorbis_parse_chain), (vorbis_parse_change_state):
* ext/vorbis/vorbisparse.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/tcp/gsttcpserversink.h:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Fix some more docs.
Added docs for vorbisdec and vorbisparse.
Fix vorbisparse.
 2006-03-01 17:39:28 +00:00
Wim Taymans 81e72d14b4 Updated/added documentation. 
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/pango/gstclockoverlay.h:
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttextrender.h:
* ext/pango/gsttimeoverlay.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* gst/audioconvert/gstaudioconvert.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gstmultifdsink.h:
Updated/added documentation.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type),
(gst_text_overlay_base_init), (gst_text_overlay_class_init),
(gst_text_overlay_init), (gst_text_overlay_set_property),
(gst_text_overlay_get_property):
Fix up properties to be enums instead of string to make bindings,
introspection and automatic GUI creation possible.
Add getters for the properties.
 2006-03-01 16:24:37 +00:00
Tim-Philipp Müller f3def77907 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun... 
Original commit message from CVS:
* gst/audioconvert/plugin.c: (plugin_init):
Register the GstAudioChannelPosition enum type with the type
system in the plugin_init function, so that it is known before
any element actually makes use of multi-channel stuff. This is
required for example if one wants to be able to deserialise/use
a caps string with channel positions before any pipeline has
been setup and started, like with gst-launch.
 2006-02-14 18:52:52 +00:00
Thomas Vander Stichele 5f83aa7dfa expand tabs 
Original commit message from CVS:
expand tabs
 2005-12-06 19:42:02 +00:00
Wim Taymans 201fd910de gst/audioconvert/gstaudioconvert.c: Post errors. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
Post errors.
 2005-12-02 10:17:35 +00:00
Thomas Vander Stichele f39b477379 borgify further clean up docs a little 
Original commit message from CVS:
borgify further
clean up docs a little
 2005-12-01 01:12:55 +00:00
Thomas Vander Stichele aa454b5383 remove sinesrc some more 
Original commit message from CVS:
remove sinesrc some more
 2005-11-23 15:36:58 +00:00
Jan Schmidt 1cc82e9138 Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) 
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)
 2005-11-21 14:29:53 +00:00
Tim-Philipp Müller 9136b3d6aa gst/audioconvert/gstaudioconvert.c: Fix typo in docs. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix typo in docs.
 2005-11-08 12:18:14 +00:00
Michael Smith 5fba886320 gst/audioconvert/audioconvert.c: When clearing an audioconvert context, set tmpbufsize to zero, so we'll allocate it ... 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
When clearing an audioconvert context, set tmpbufsize to zero, so
we'll allocate it again later if required.
This fixes audioconvert re-negotiating formats, which previously
segfaulted with a NULL destination buffer.
 2005-10-26 16:26:44 +00:00
Tim-Philipp Müller 19efac956e gst/audioconvert/: And bye bye buffer-frames-convert 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/bufferframesconvert.c:
* gst/audioconvert/plugin.c: (plugin_init):
* gst/audioconvert/plugin.h:
And bye bye buffer-frames-convert
 2005-10-19 17:50:17 +00:00
Wim Taymans fc8ce00673 Bye bye buffer-frames. 
Original commit message from CVS:
* check/elements/audioconvert.c:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_identification_packet), (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (raw_caps_factory):
* gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
* gst/volume/gstvolume.c:
Bye bye buffer-frames.
 2005-10-19 17:02:46 +00:00
Thomas Vander Stichele 4f8f42b0b6 restructure configure.ac, use correct libtool LDFLAGS, fix up defines 
Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
 2005-10-16 13:54:44 +00:00
Michael Smith bd3cff086d - Don't use non-portable LL suffix on constants, since MSVC doesn't allow them. These constants all fit into ints any... 
Original commit message from CVS:
- Don't use non-portable LL suffix on constants, since MSVC doesn't allow
them. These constants all fit into ints anyway.

- Continue to hate nano.
 2005-10-13 11:36:14 +00:00
Wim Taymans 5b3f6be65c gst/audioconvert/: Alloc temp storage somewhere else where we can do it more portable. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
Alloc temp storage somewhere else where we can do it more
portable.
 2005-10-10 13:45:39 +00:00
Thomas Vander Stichele d9d1b4a934 some documentation for audioconvert 
Original commit message from CVS:
some documentation for audioconvert
 2005-09-23 14:41:31 +00:00
Wim Taymans b6bc76642a gst/audioconvert/gstaudioconvert.c: And enable 24 bits mode as well.. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
And enable 24 bits mode as well..
 2005-09-15 13:52:27 +00:00
Tim-Philipp Müller 32f976bfea gst/audioconvert/Makefile.am: Audioconvert derives from GstBaseTransform and should link to the library with our base... 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
 2005-09-13 13:52:59 +00:00
Wim Taymans 1237e1e701 gst/audioconvert/audioconvert.*: Cleanups, speedups, simplifications, added back support for 24 bits. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
Cleanups, speedups, simplifications, added back support
for 24 bits.
 2005-09-12 11:38:05 +00:00
Jan Schmidt 0f4fa24d8e check/: Add extra tests for basetransform based components. 
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.
 2005-09-09 17:53:47 +00:00
Thomas Vander Stichele a23795d4d3 disable 24 bit until it gets fixed 
Original commit message from CVS:
disable 24 bit until it gets fixed
 2005-09-02 23:16:15 +00:00
Andy Wingo 6665c3084c All plugins updated for element state changes. 
Original commit message from CVS:
2005-09-02  Andy Wingo  <wingo@pobox.com>

* All plugins updated for element state changes.
 2005-09-02 15:43:18 +00:00
Andy Wingo c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL. 
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
 2005-08-28 17:52:45 +00:00
Wim Taymans b6c368ce67 gst/audioconvert/audioconvert.c: Cleanups. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
Cleanups.
 2005-08-26 18:57:30 +00:00
Wim Taymans ddec57c089 gst/audioconvert/audioconvert.c: More elegant and working temp buffer selection algo. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
 2005-08-26 18:43:02 +00:00
Wim Taymans 123aa7de1a gst/audioconvert/audioconvert.c: Use realloc else we lose our original data. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
Use realloc else we lose our original data.
 2005-08-26 17:46:45 +00:00
Wim Taymans 98fbd82d1c gst/audioconvert/: Oops, allocate enough space to perform the channel mix. 
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
Oops, allocate enough space to perform the channel mix.
 2005-08-26 17:30:41 +00:00
Wim Taymans ceb84de916 gst/audioconvert/: Cleanups, librarify a bit, optimize, better negotiation and more. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
(gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
(gst_audio_convert_transform_ip), (gst_audio_convert_transform):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_fill_identical),
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
(gst_channel_mix_mix):
* gst/audioconvert/gstchannelmix.h:
Cleanups, librarify a bit, optimize, better negotiation and more.
 2005-08-26 15:43:56 +00:00
Thomas Vander Stichele 43332aed85 plug some leaks 
Original commit message from CVS:
plug some leaks
 2005-08-25 17:32:34 +00:00
Thomas Vander Stichele 41a43b86a8 port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes 
Original commit message from CVS:
port audioconvert to basetransform
fix ffmpegcsp and videoscale for basetransform changes
 2005-08-24 13:32:52 +00:00
Thomas Vander Stichele 3e589b0977 rename plugin 
Original commit message from CVS:
rename plugin
 2005-08-15 14:15:40 +00:00
Thomas Vander Stichele e571f069d1 renamed to actual element names, so much nicer to look at 
Original commit message from CVS:

* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
 2005-08-05 18:51:29 +00:00
Wim Taymans a6d89f51fa gst/audioconvert/gstaudioconvert.c: Convert me to BaseTransform!! help.. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
Convert me to BaseTransform!! help..
 2005-07-29 17:07:39 +00:00
Wim Taymans 567802ca2c gst/audioconvert/gstaudioconvert.c: Timestamp buffers correctly. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
Timestamp buffers correctly.

* gst/playback/gstplaybin.c: (gen_video_element):
Make internal fakesink silent.
 2005-07-16 13:58:21 +00:00
Thomas Vander Stichele 1ea0574af4 make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be added manually to each Makefile.am so we are sure it goes 
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
 2005-07-13 17:58:07 +00:00
Wim Taymans a46a991d26 ext/theora/theoradec.c: Prepare for better timestamp fix later. 
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.

* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first

* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.

* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
 2005-07-06 15:14:38 +00:00
Andy Wingo 3a7f5a2e06 gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate): No refcount leakage. 
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
 2005-07-04 10:40:17 +00:00
Andy Wingo 1f40231de5 configure.ac: Enable -Werror. 
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* configure.ac: Enable -Werror.

* ext/theora/theoradec.c (theora_dec_src_getcaps):
* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate):
* gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int)
(gst_audio_convert_fixate):
* gst/sine/gstsinesrc.c (gst_sinesrc_src_fixate)
(gst_sinesrc_create): Fixate func changes.

* sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
(gst_ximagesink_buffer_alloc): Unused var.
 2005-07-04 10:37:25 +00:00
Andy Wingo d0bc038021 *.c: Don't cast to GstObject before reffing/unreffing. 
Original commit message from CVS:
2005-06-28  Andy Wingo  <wingo@pobox.com>

* *.c: Don't cast to GstObject before reffing/unreffing.
 2005-06-28 10:16:13 +00:00
Andy Wingo 55d437af1c gst/: Ghost pad API fixes. 
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/gconf/gconf.c:
* gst/playback/test.c:
* gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
fixes.

* gst/audioconvert/gstaudioconvert.c: RPAD fixes.

* ext/theora/theoraenc.c (theora_enc_chain):
* ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.

* ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
RealPad.
 2005-06-08 22:18:05 +00:00
Wim Taymans 6db639089a Cleanups and buffer alloc. 
Original commit message from CVS:
* ext/ogg/README:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_clear_chains):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_data_packet):
* gst/audioconvert/bufferframesconvert.c:
(buffer_frames_convert_chain):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
(gst_ffmpegcsp_chain):
* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
(gst_videorate_getcaps), (gst_videorate_setcaps),
(gst_videorate_event), (gst_videorate_chain):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_activate),
(gst_videotestsrc_src_query), (gst_videotestsrc_loop):
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
Cleanups and buffer alloc.
 2005-06-02 09:46:40 +00:00
Wim Taymans 5474600d4f gst-libs/gst/audio/: Various small cleanups. 
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.
 2005-05-25 19:52:14 +00:00
Ronald S. Bultje 78016b40cf gst/audioconvert/gstaudioconvert.c: Implement instant setup switching. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_chain), (gst_audio_convert_link_src),
(gst_audio_convert_setcaps):
Implement instant setup switching.
 2005-05-23 17:28:02 +00:00
Wim Taymans 04fa67937d Leak fixes in oggdemux. 
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
(gst_ogg_demux_chain_unlocked):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_change_state):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_init),
(gst_ffmpegcsp_bufferalloc), (gst_ffmpegcsp_chain),
(gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
(gst_ffmpegcsp_get_property):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_imagepool_clear),
(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_free),
(gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id):
Leak fixes in oggdemux.
Some cleanups in audioconvert.
Make passthrough work along with buffer_alloc etc.
Make buffer_alloc and buffer recycling actually work in
xvimagesink.
 2005-05-17 17:41:32 +00:00
David Schleef d90ee5bfa3 Port from GstData to GstMiniObject. 
Original commit message from CVS:
Port from GstData to GstMiniObject.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
(gst_ogg_mux_collected):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_comment_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_set_header_on_caps), (theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_comment_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/volume/gstvolume.c: (volume_transform):
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
(gst_ximagesink_buffer_alloc):
* sys/ximage/ximagesink.h:
 2005-05-16 15:35:52 +00:00
Wim Taymans 8aeaf8ed14 Make caps writable before writing to it. 
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
Make caps writable before writing to it.
Fix negotiation in audioconvert some more.
 2005-05-09 17:07:27 +00:00
Wim Taymans 3ec8704c9e Fixed negotiation wrt _peer_get_caps() 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
* gst/sine/Makefile.am:
* gst/sine/gstsinesrc.c: (gst_sinesrc_get_type),
(gst_sinesrc_class_init), (gst_sinesrc_init),
(gst_sinesrc_src_fixate), (gst_sinesrc_setcaps),
(gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start),
(gst_sinesrc_update_freq):
* gst/sine/gstsinesrc.h:
* gst/tcp/gstmultifdsink.c:
* sys/xvimage/xvimagesink.c:
Fixed negotiation wrt _peer_get_caps()
Some cleanups.
 2005-05-06 17:13:49 +00:00
Andy Wingo 790c059867 gst/: Some GCC4 fixes 
Original commit message from CVS:
2005-05-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer), (gst_vorbis_tag_chain):
* gst/adder/gstadder.h:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_one_other):
* gst/audiorate/gstaudiorate.c: (gst_audiorate_setcaps),
(gst_audiorate_init), (gst_audiorate_chain):
* gst/playback/gstplaybasebin.c: (setup_source):
* gst/playback/test3.c: (update_scale):
Some GCC4 fixes

* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po: Foo
 2005-05-05 14:57:20 +00:00
Wim Taymans 3e5febcdd1 gst/: Don't ignore _push() return values. 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_change_state), (gst_audio_convert_channels):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
(gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
(gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
(gst_videotestsrc_init), (gst_videotestsrc_loop):
Don't ignore _push() return values.
Make sure no processing is done when shutting down.
Videotestsrc pad activation fix.
 2005-05-05 09:49:08 +00:00
Thomas Vander Stichele 37822dc3fb Fix part of the build. Come on guys, autogen didn't even work :) 
Original commit message from CVS:
Fix part of the build.  Come on guys, autogen didn't even work :)
 2005-04-25 13:29:40 +00:00
David Schleef 129c7e8af1 configure.ac: Remove idct and resample libs 
Original commit message from CVS:
* configure.ac: Remove idct and resample libs
* gst-libs/gst/Makefile.am: same
Remove usage of gst_library_load():
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/libvisual/visual.c: (plugin_init):
* ext/ogg/gstogg.c: (plugin_init):
* ext/theora/theora.c: (plugin_init):
* ext/vorbis/vorbis.c: (plugin_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
* gst/audioscale/gstaudioscale.c:
* gst/adder/gstadder.c: (plugin_init):
* gst/audioconvert/plugin.c: (plugin_init):
* sys/ximage/ximagesink.c: (plugin_init):
* sys/xvimage/xvimagesink.c: (plugin_init):
* gst/tcp/gsttcpplugin.c: (plugin_init):
Link plugins against libraries:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/audioconvert/Makefile.am:
Create proper libraries:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Move resample library to audioscale plugin directory:
* gst-libs/gst/resample/Makefile.am:
* gst-libs/gst/resample/README:
* gst-libs/gst/resample/dtof.c:
* gst-libs/gst/resample/dtos.c:
* gst-libs/gst/resample/functable.c:
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c:
* gst-libs/gst/resample/resample.h:
* gst-libs/gst/resample/resample.vcproj:
* gst-libs/gst/resample/test.c:
* gst/audioscale/Makefile.am:
* gst/audioscale/README:
* gst/audioscale/dtof.c:
* gst/audioscale/dtos.c:
* gst/audioscale/functable.c:
* gst/audioscale/private.h:
* gst/audioscale/resample.c:
* gst/audioscale/resample.h:
* gst/audioscale/test.c:
Move tagedit library to gst-libs:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gsttagediting.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst/tags/Makefile.am:
* gst/tags/gstid3tag.c:
* gst/tags/gstvorbistag.c:
Fix for core changes:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
(gst_sinesrc_getrange):
 2005-04-25 00:23:06 +00:00
Wim Taymans 468d6d4326 gst/audioconvert/: Link against audio libs. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_channels):
Link against audio libs.
Fix audio convert plugin.
 2005-04-20 09:45:33 +00:00
Wim Taymans 1dae961cbf Plugin port to 0.9, ogg/theora playback should work in the seek example now. 
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
 2005-03-31 09:43:49 +00:00
Tim-Philipp Müller 797a080e39 Don't leak othercaps in link function (fixes #167878) 
Original commit message from CVS:
Don't leak othercaps in link function (fixes #167878)
 2005-02-19 20:01:36 +00:00
Benjamin Otte c196377ff6 gst/audioconvert/gstaudioconvert.c: create channel conversion matrix when linking 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(gst_audio_convert_channels):
create channel conversion matrix when linking
* gst/audioconvert/.cvsignore:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/channelmixtest.c: (main):
add (ugly) test that ensures stereo <=> mono conversion works
correctly
 2005-02-13 17:39:22 +00:00
Benjamin Otte f9cf10c748 gst/audioconvert/gstchannelmix.h: include missing header file 
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.h:
include missing header file
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_compatible):
use same sign for both channels when converting to/from compatible
channel. Previously used different signs made the signals cancel
each other out and appear like silence. (fixes #167269)
 2005-02-13 16:10:16 +00:00
Jan Schmidt 83e3fe189c configure.ac: Add dvdlpcmdec 
Original commit message from CVS:

* configure.ac:
Add dvdlpcmdec

* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_reset),
(free_all_buffers), (gst_mpeg2dec_alloc_buffer):
Don't push buffers if the src pad isn't negotiated yet.

* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format),
(gst_audio_convert_buffer_from_default_format):
Add support for 24-bit width.

* gst/dvdlpcmdec/.cvsignore:
* gst/dvdlpcmdec/Makefile.am:
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_get_type),
(gst_dvdlpcmdec_base_init), (gst_dvdlpcmdec_class_init),
(gst_dvdlpcm_reset), (gst_dvdlpcmdec_init), (gst_dvdlpcmdec_link),
(gst_dvdlpcmdec_chain), (gst_dvdlpcmdec_change_state),
(plugin_init):
* gst/dvdlpcmdec/gstdvdlpcmdec.h:
New decoder for rearranging DVD LPCM into our audio/x-raw-int
format. Needs support for the channels maps if someone can find
a DVD LPCM track with > 2 channels.

* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_handle_dvd_event),
(gst_dvd_demux_send_discont), (gst_dvd_demux_handle_discont),
(gst_dvd_demux_get_audio_stream), (gst_dvd_demux_process_private):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_init_stream),
(gst_mpeg_demux_send_subbuffer), (gst_mpeg_demux_handle_src_query):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_reset),
(gst_mpeg_parse_parse_packhead), (gst_mpeg_parse_loop),
(gst_mpeg_parse_get_rate), (gst_mpeg_parse_convert_src),
(gst_mpeg_parse_handle_src_query),
(gst_mpeg_parse_handle_src_event):
Use audio/x-dvd-lpcm for LPCM output.
Add DTS output.
 2005-02-08 11:08:15 +00:00
Andy Wingo 9c1c858d24 gst/audioconvert/bufferframesconvert.c 
Original commit message from CVS:
2005-02-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate): New function, fixates to 256
frames per buffer by default. (Much better than 1.)
(buffer_frames_convert_init): Set the fixate function for both src
and sink pad.
(buffer_frames_convert_link): After success setting nonfixed caps,
get the negotiated caps so we can know how many buffer-frames it
will be. No idea how this worked at all before.
 2005-02-04 15:40:38 +00:00
Ronald S. Bultje 97dd32f748 gst/audioconvert/gstaudioconvert.c: The return value of fixate_to does not imply that the requested value was set, so... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
The return value of fixate_to does not imply that the requested
value was set, so don't assume.
 2005-01-07 18:17:52 +00:00
Thomas Vander Stichele c1e1206414 fix for glib < 2.4 
Original commit message from CVS:
fix for glib < 2.4
 2004-12-24 15:12:56 +00:00
Benjamin Otte 05103c18ca gst/audioconvert/gstchannelmix.c: more overwriting protection due to modifying channels one by one instead of all at ... 
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
more overwriting protection due to modifying channels one by one
instead of all at once
 2004-11-28 20:41:23 +00:00
Ronald S. Bultje 836ea71ea1 gst/audioconvert/gstchannelmix.c: Normalize using absolute values. 
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_normalize):
Normalize using absolute values.
 2004-11-28 19:22:07 +00:00
Benjamin Otte 110fd90222 gst/audioconvert/gstchannelmix.c: walk the samples backwards if out_channels > in_channels so we don't overwrite data 
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
walk the samples backwards if out_channels > in_channels so we don't
overwrite data
 2004-11-28 18:26:36 +00:00
Ronald S. Bultje 4fdf8d5b16 gst/audioconvert/: Implement a channel mixer. 
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_channels):
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_unset_matrix),
(gst_audio_convert_fill_identical),
(gst_audio_convert_fill_compatible),
(gst_audio_convert_detect_pos), (gst_audio_convert_fill_one_other),
(gst_audio_convert_fill_others),
(gst_audio_convert_fill_normalize),
(gst_audio_convert_fill_matrix), (gst_audio_convert_setup_matrix),
(gst_audio_convert_passthrough), (gst_audio_convert_mix):
* gst/audioconvert/gstchannelmix.h:
Implement a channel mixer.
 2004-11-28 16:09:13 +00:00
Ronald S. Bultje 1de3e19fdb ext/a52dec/gsta52dec.c: Don't do sample adjusting anymore, we use float audio now. 
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_loop),
(gst_a52dec_change_state):
Don't do sample adjusting anymore, we use float audio now.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
Don't fixate to non-existing properties.
 2004-11-27 20:22:42 +00:00
Christophe Fergeau 39f436c1a0 gst/audioconvert/gstaudioconvert.c: call parent dispose method 
Original commit message from CVS:
2004-11-27  Christophe Fergeau  <teuf@gnome.org>

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
call parent dispose method
 2004-11-27 14:41:51 +00:00
Ronald S. Bultje 3a0a2898af Surround sound support. 
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
 2004-11-25 20:36:29 +00:00
Ronald S. Bultje b463251864 gst/audioconvert/gstaudioconvert.c: Really don't touch read-only buffers (#156563). 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format):
Really don't touch read-only buffers (#156563).
 2004-10-29 12:48:45 +00:00
Iain Holmes 02be6646cc Free the caps used for the try 
Original commit message from CVS:
Free the caps used for the try
 2004-09-16 11:34:50 +00:00
David Schleef 78deea7e4a configure.ac: remove NASM check, since we don't use it. Update dirac check to 0.4 
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it.  Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
 2004-09-15 19:29:24 +00:00
Steve Lhomme 7b86915b8a more working plugins 
Original commit message from CVS:
more working plugins
 2004-07-27 21:41:30 +00:00
Steve Lhomme a6c379fd7a rename GStreamer-0.8.lib to libgstreamer.lib 
Original commit message from CVS:
rename GStreamer-0.8.lib to libgstreamer.lib
 2004-07-27 09:57:33 +00:00
Steve Lhomme 6c8a44d0c9 avoid problems with math.h, fix release dependancy 
Original commit message from CVS:
avoid problems with math.h, fix release dependancy
 2004-07-27 09:48:51 +00:00
Steve Lhomme 17615126c2 more plugins supported under windows 
Original commit message from CVS:
more plugins supported under windows
 2004-07-26 13:20:10 +00:00
Benjamin Otte d5f5085677 gst/audioconvert/gstaudioconvert.c: don't enfore negotiation from source side, it breaks sinesrc ! audioconvert ! oss... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
don't enfore negotiation from source side, it breaks
sinesrc ! audioconvert ! osssink
 2004-07-23 17:40:16 +00:00
Andy Wingo b028fc77a7 gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For float, "any" caps -> buffer_frames=[0,MAX]. 
Original commit message from CVS:
2004-07-11  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For
float, "any" caps -> buffer_frames=[0,MAX].

* gst/interleave/interleave.c (interleave_getcaps): Seems the core
doesn't intersect our caps with the template any more. Do it
ourselves.
(interleave_buffered_loop): Use g_newa instead of malloc/free.
 2004-07-11 11:21:56 +00:00
Thomas Vander Stichele c750d9bd49 don't assert in state change 
Original commit message from CVS:
don't assert in state change
 2004-07-09 10:56:51 +00:00
Benjamin Otte b95a7dca6a gst/audioconvert/gstaudioconvert.c: fixate nicely even when the peer is not negotiating 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate nicely even when the peer is not negotiating
 2004-05-26 14:47:23 +00:00
Benjamin Otte 4f845c50ce gst/audioconvert/gstaudioconvert.c: make sure we don't allow depth > width 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
make sure we don't allow depth > width
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate endianness to G_BYTE_ORDER as default
* gst/audioscale/gstaudioscale.c:
we don't handle another endianness as host-endianness
 2004-05-25 20:14:10 +00:00
Benjamin Otte 2e050e0378 ext/vorbis/oggvorbisenc.c: properly fail when we can't setup the vorbis encoder due to unsupported settings 
Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
(gst_oggvorbisenc_setup):
properly fail when we can't setup the vorbis encoder due to
unsupported settings
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
(gst_vorbisenc_setup):
same
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
fix case where warnings occured when one pad was unlinked while the
other's link function was called
 2004-05-24 19:19:29 +00:00
Stéphane Loeuillet 1f1a7cbe84 first batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc 
Original commit message from CVS:

first batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
 2004-05-21 22:39:30 +00:00
Thomas Vander Stichele cba2022045 gst/audioconvert/gstaudioconvert.c: refactor/comment code 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
refactor/comment code
 2004-05-03 13:25:22 +00:00
Benjamin Otte 1cd4212d70 gst/audioconvert/gstaudioconvert.c: fix memleak 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
fix memleak
 2004-04-25 17:56:11 +00:00
Benjamin Otte 26cc5e8768 ext/mad/gstid3tag.c: remove leftover g_print 
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init):
remove leftover g_print
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
don't try setting only a subset of the caps. We don't want to kill
autoplugging on purpose
 2004-04-20 15:51:48 +00:00
Thomas Vander Stichele d32724fb41 add debugging 
Original commit message from CVS:
add debugging
 2004-04-14 16:09:10 +00:00
Benjamin Otte 50d120f1ba ext/gnomevfs/gstgnomevfssink.c: fix erase signal - if any handler returns false the file will not be overwritten. If ... 
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(_gst_boolean_allow_overwrite_accumulator),
(gst_gnomevfssink_class_init):
fix erase signal - if any handler returns false the file will not be
overwritten. If no handler is connected, the file will not be
overwritten either.
renamed signal to "allow-overwrite"
* ext/mad/gstid3tag.c: (tag_list_to_id3_tag_foreach):
free string when adding it to ID3 failed
* ext/vorbis/vorbisdec.c: (vorbis_dec_event):
unref event when done
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
free caps
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_type_find):
fix invalid read
 2004-04-09 18:55:10 +00:00
Andy Wingo 92fe387eea gst/audioconvert/bufferframesconvert.c: New element to convert buffer-frames for float streams. Not working nicely yet. 
Original commit message from CVS:
2004-04-09  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c: New element to convert
buffer-frames for float streams. Not working nicely yet.
* gst/audioconvert/plugin.h:
* gst/audioconvert/plugin.c: New files.
* gst/audioconvert/Makefile.am: Build the new files.
* gst/audioconvert/gstaudioconvert.c: Initialize via plugin.[ch].
 2004-04-09 12:39:30 +00:00
Benjamin Otte 0db7a00219 gst/audioconvert/gstaudioconvert.c: advertise buffer-frames correctly on sinkpads 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps):
advertise buffer-frames correctly on sinkpads
 2004-04-05 13:18:56 +00:00
Benjamin Otte 87ffc58ab9 gst/audioconvert/gstaudioconvert.c: add a fixation function that pretty much does the right thing (fixes #137556) 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(_fixate_caps_to_int), (gst_audio_convert_fixate):
add a fixation function that pretty much does the right thing (fixes
#137556)
 2004-03-21 02:54:37 +00:00
Thomas Vander Stichele f83cb187de don't mix tabs and spaces 
Original commit message from CVS:
don't mix tabs and spaces
 2004-03-15 19:32:28 +00:00
Thomas Vander Stichele 4df3f18839 gst-indent 
Original commit message from CVS:
gst-indent
 2004-03-14 22:34:34 +00:00
Colin Walters e93d93afdf gst/audioconvert/gstaudioconvert.c: Fix typo in width 8 conversion. 
Original commit message from CVS:
2004-03-09  Colin Walters  <walters@verbum.org>

* gst/audioconvert/gstaudioconvert.c: Fix typo in width 8
conversion.
 2004-03-10 04:01:50 +00:00
Benjamin Otte c6b75500be add some 'what's this element and what is it not' doc 
Original commit message from CVS:
add some 'what's this element and what is it not' doc
 2004-03-06 15:31:25 +00:00
Benjamin Otte 33f79a881e gst/audioconvert/gstaudioconvert.c: do conversions from/to float correctly, fix some caps nego errors, export correct... 
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_buffer_from_default_format):
do conversions from/to float correctly, fix some caps nego errors,
export correct supported caps in template and getcaps, use correct
caps in try_set_caps functions
 2004-03-06 13:26:12 +00:00
David Schleef f0365ebe22 ext/aalib/gstaasink.c: Add fixate function. (bug #131128) 
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_fixate), (gst_aasink_init):
Add fixate function. (bug #131128)
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_fixate):  Add fixate function.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Fix attempt to print a non-pointer using GST_PTR_FORMAT.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt):
Fix missing break that was causing ulaw to be interpreted as
raw int.
 2004-03-06 04:51:15 +00:00