Commit graph

345 commits

Author SHA1 Message Date
Tim-Philipp Müller
4936d6ba02 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
Remove compatibility cruft for prehistoric GLib versions.
2007-01-18 16:23:35 +00:00
Thomas Vander Stichele
95ada43982 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
Original commit message from CVS:
* configure.ac:
split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
so that GST_BASE_CFLAGS can go inbetween them, making sure
we use uninstalled gst-libs headers
* docs/libs/Makefile.am:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
* tests/icles/Makefile.am:
adapt
2007-01-04 12:49:48 +00:00
Josep Torre Valles
4de10dacb6 ext/gnomevfs/: Fix URI interface implementation return type.
Original commit message from CVS:
2006-10-10  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

Patch by: Josep Torre Valles <josep@fluendo.com>

* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
2006-10-10 12:49:03 +00:00
Tim-Philipp Müller
f306655319 gst/: A few array const-ifications.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
* gst/videotestsrc/videotestsrc.h:
A few array const-ifications.
2006-09-23 15:24:55 +00:00
Stefan Kost
f2fbfdc124 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(gst_audio_convert_transform_caps):
Get structure-name just once.
2006-09-17 11:24:21 +00:00
Stefan Kost
c2d7af84c1 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_unit_size), (set_structure_widths):
Lower debug, use g_assert in _get_unit_size
* gst/audioresample/gstaudioresample.c:
(audioresample_get_unit_size):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
use g_assert in _get_unit_size
2006-08-20 13:05:44 +00:00
Tim-Philipp Müller
8a223b920d gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
Float caps shouldn't have a "signed" field.
2006-08-10 13:01:31 +00:00
Michael Smith
fc30337a99 gst/audioconvert/audioconvert.c: More correct float->int conversion.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float):
More correct float->int conversion.
2006-06-02 14:19:18 +00:00
Stefan Kost
131fb86b4b Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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
2006-06-01 19:19:51 +00:00
Michael Smith
8e09be1bd9 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(append_with_other_format), (set_structure_widths),
(gst_audio_convert_transform_caps):
Patch from #341562: give more specific audio caps in get_caps, so
that basetransform  can make better decisions on what caps to
negotiate.
2006-05-29 11:04:48 +00:00
Michael Smith
400ade1e8f gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix #341696: crash when mixing L+R+C to mono or stereo.
* tests/check/Makefile.am:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
(audioconvert_suite):
Add test for the above, including some generic framework bits for
testing multichannel things.
2006-05-14 18:15:17 +00:00
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
David Schleef
befdae8cda ext/faad/gstfaad.c: Fix negotiation.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):  Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c:  Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c:  Add glib header
* pkgconfig/gstreamer-play.pc.in:  Depends on gst-interfaces.
2004-03-06 00:42:20 +00:00
Benjamin Otte
043693d8d9 gst/audioconvert/gstaudioconvert.c: convert channels correctly. convert correctly to unsigned.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
convert channels correctly. convert correctly to unsigned.
2004-03-05 21:05:26 +00:00
Benjamin Otte
02c11b879e gst/audioconvert/gstaudioconvert.c: make float=>int conversion work correctly even in cornercases.
Original commit message from CVS:
2004-03-05  Benjamin Otte  <otte@gnome.org>

* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format):
make float=>int conversion work correctly even in cornercases.
2004-03-04 23:30:29 +00:00
Thomas Vander Stichele
8008cf22a4 PTR fix
Original commit message from CVS:
PTR fix
2004-02-25 18:02:19 +00:00
Thomas Vander Stichele
d31fb663bd assorted debug/warning fixes
Original commit message from CVS:
assorted debug/warning fixes
2004-02-25 17:45:54 +00:00
Andy Wingo
cd8976a9eb gst/interleave/interleave.c (interleave_buffered_loop): Always push only when channel->buffer is NULL. Prevents segfa...
Original commit message from CVS:
2004-02-25  Andy Wingo  <wingo@pobox.com>

* gst/interleave/interleave.c (interleave_buffered_loop): Always
push only when channel->buffer is NULL. Prevents segfaults doing
the state change after a nonlocal exit, like a scheme exception.

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
Handle the case where the intersected caps is empty.
2004-02-25 13:25:44 +00:00
Benjamin Otte
355318b59d gst/audioconvert/gstaudioconvert.c: set rank to PRIMARY
Original commit message from CVS:
2004-02-22  Benjamin Otte  <otte@gnome.org>

reported by: Stefan Kost <kost@imn.htwk-leipzig.de>

* gst/audioconvert/gstaudioconvert.c: (plugin_init):
set rank to PRIMARY
* gst/volume/gstvolume.c: (plugin_init):
set rank to NONE
fixes #134960

2004-02-22   Julio M. Merino Vidal <jmmv@menta.net>

reviewed by Benjamin Otte  <otte@gnome.org>

* ext/flac/gstflacenc.c: (gst_flacenc_chain):
escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
2004-02-22 15:31:30 +00:00
Andy Wingo
ce89f16818 gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and interleave respectively.
Original commit message from CVS:
2004-02-20  Andy Wingo  <wingo@pobox.com>

* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.

* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.

* configure.ac: Remove intfloat and oneton, add interleave.

* ext/sndfile/gstsf.c: Handle events better.

* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
2004-02-20 14:17:57 +00:00
Thomas Vander Stichele
0f003d87b3 throw error instead of assertion
Original commit message from CVS:
throw error instead of assertion
2004-02-16 11:45:32 +00:00
David Schleef
8fe7678826 gst/audioconvert/gstaudioconvert.c: Use gst_pad_try_set_caps_nonfixed().
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Use gst_pad_try_set_caps_nonfixed().
2004-01-27 09:05:22 +00:00
David Schleef
e4593b1536 gst/ac3parse/gstac3parse.c: update to checklist 5
Original commit message from CVS:
* gst/ac3parse/gstac3parse.c: update to checklist 5
* gst/adder/gstadder.c: rewrite negotiation.  update to checklist 5
* gst/audioconvert/gstaudioconvert.c: update to checklist 5
* gst/audioscale/gstaudioscale.c: same
* gst/auparse/gstauparse.c: same
* gst/avi/gstavidemux.c: same
2004-01-27 09:00:01 +00:00
Benjamin Otte
b552a2d9f0 gst-libs/gst/audio/audio.h: remove buffer-frames from audio caps
Original commit message from CVS:
2004-01-26  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
2004-01-26 03:54:21 +00:00
David Schleef
89303c580f ext/esd/esdsink.c: Remove property that handles osssink fallback.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_class_init): Remove property
that handles osssink fallback.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_getcaps):
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Add audio/x-qdm2 for QDM2 audio.
* gst/sine/gstsinesrc.c: (gst_sinesrc_get):
* gst/sine/gstsinesrc.h: Add example of how to implement tags.
* gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
Decrease minimum size to 16x16.
* gst/wavparse/gstwavparse.c:
Convert disabled pad template caps to new caps.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_chain): Throw element error when display cannot
be opened.  Increase minimum framerate to 1.0.  Check the data
free function on a buffer to make sure it is the type we expect
before manipulating it.
2004-01-15 21:05:17 +00:00
Thomas Vander Stichele
2a415f1abd fix up audioconvert caps nego, remove float stuff, remove rate stuff gst-launch-0.7 -v sinesrc ! audioconvert ! audi...
Original commit message from CVS:

fix up audioconvert caps nego, remove float stuff, remove rate stuff
gst-launch-0.7  -v sinesrc ! audioconvert ! audio/x-raw-int,rate=23000 ! wavenc ! filesink location=test.wav now writes a completely useless 23000 Hz wave file
2004-01-12 19:46:45 +00:00
David Schleef
61763a83f0 gst/audioconvert/gstaudioconvert.c: Test that pad is negotiated before getting its caps.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Test that pad is negotiated before getting its caps.
2004-01-12 18:59:57 +00:00
David Schleef
e095323df1 Negotiation fixes.
Original commit message from CVS:
Negotiation fixes.
2003-12-31 08:02:04 +00:00
David Schleef
3b60021408 Merge CAPS branch
Original commit message from CVS:
Merge CAPS branch
2003-12-22 01:47:09 +00:00
Andy Wingo
80fece4f4b remove copyright field from plugins
Original commit message from CVS:
remove copyright field from plugins
2003-12-04 10:37:39 +00:00
Leif Johnson
736153ab06 + checking in plugin category changes
Original commit message from CVS:
+ checking in plugin category changes
2003-11-16 22:02:21 +00:00
Iain Holmes
3d97918694 Audioconvert - Check!
Original commit message from CVS:
Audioconvert - Check!
Updated for new stuff
2003-11-01 11:41:42 +00:00
Andy Wingo
dc35dbb595 /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
Original commit message from CVS:
/GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
2003-10-08 16:08:22 +00:00
Thomas Vander Stichele
453e9b8871 reverting error patch before making a branch.
Original commit message from CVS:
reverting error patch before making a branch.
2003-09-16 10:00:02 +00:00
Benjamin Otte
d26698b5a1 converted gst_element_error to new format in gst/ - gettext pending
Original commit message from CVS:
converted gst_element_error to new format in gst/ - gettext pending
2003-09-15 00:34:44 +00:00
David Schleef
0228717f04 Remove redundant plugindir definition
Original commit message from CVS:
Remove redundant plugindir definition
2003-08-10 00:01:58 +00:00
Leif Johnson
2ad6bd23cf + some whitespace changes + adding dummy definitions to prepare for float caps
Original commit message from CVS:
+ some whitespace changes
+ adding dummy definitions to prepare for float caps
2003-07-19 22:58:41 +00:00
Ronald S. Bultje
b005531324 New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
2003-07-06 20:49:52 +00:00
Benjamin Otte
ffa5706370 compatibility fix for new GST_DEBUG stuff.
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h

I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
2003-06-29 19:46:12 +00:00
David Schleef
b036fe4fda fix: width is in bytes, not bits. Remove incorrect assertion.
Original commit message from CVS:
fix: width is in bytes, not bits.  Remove incorrect assertion.
2003-06-10 06:32:59 +00:00
Benjamin Otte
b3c728ed0d *_is_writeable => *_is_writable (spelling)
Original commit message from CVS:
*_is_writeable => *_is_writable (spelling)
2003-04-16 18:36:29 +00:00
Benjamin Otte
333099fdeb change *_is_readonly to *_is_writeable. Hope the name stays now...
Original commit message from CVS:
change *_is_readonly to *_is_writeable. Hope the name stays now...
2003-04-15 21:35:08 +00:00
Benjamin Otte
f38ca0a7f4 - revert change to use a newly added gst_buffer_is_readonly (which wasn't newly added before)
Original commit message from CVS:
- revert change to use a newly added gst_buffer_is_readonly (which wasn't newly added before)
- walk buffer backwards when it might be possible that data is read out of overwritten parts when going forwards
2003-04-15 19:10:14 +00:00
Colin Walters
804522784e Change agressive -> aggressive.
Original commit message from CVS:
Change agressive -> aggressive.
2003-04-15 03:39:22 +00:00
Colin Walters
8aa9339f50 Initialize various variables so gcc won't complain.
Original commit message from CVS:
Initialize various variables so gcc won't complain.

Use GST_BUFFER_FLAG_IS_SET instead of unknown function gst_buffer_is_readonly.
2003-04-15 03:19:08 +00:00
Benjamin Otte
d41620bc75 Added initial version of audioconvert, a generic converter of integer audio/raw formats.
Original commit message from CVS:
Added initial version of audioconvert, a generic converter of integer audio/raw formats.
It currently supports conversion of
- channels (mono/stereo only, until someone tells me how to mix other channels)
- endianness (little/bi endian)
- signedness
- width (8, 1, 24 and 32 bits)
- depth (1 - width bits)
missing:
- enough testing (I intend to write a testsuite for this, but that's pending)
- samplerate conversion
- other goodies like format conversion etc
Expect bugs when using it.

problems this should solve:
- encoding wav files on big endian machines
- goom working with mono audio files in gst-player
- Iain's soundcard (that one is a problem in itself)
- complaints about missing conversion
- too many age old, nearly unmaintained plugins (stereo2mono etc.)
Have fun.
2003-04-14 01:20:30 +00:00