Commit graph

117 commits

Author SHA1 Message Date
Sebastian Dröge
dbb857b93b gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
Sebastian Dröge
84c824b952 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
2007-04-24 18:58:25 +00:00
René Stadler
6ac8ff9ec3 with some minor changes
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
2007-03-29 18:42:34 +00:00
Michael Smith
e1544977a6 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
2007-03-27 11:31:17 +00:00
Michael Smith
3bc107dd77 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
2007-03-16 17:29:09 +00:00
Michael Smith
5759241eb4 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
2007-03-16 16:42:23 +00:00
Thomas Vander Stichele
e81b9ec719 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
fix error category and translatable string
2007-03-09 12:22:53 +00:00
Stefan Kost
2c67c89457 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes #339837 even more.
2007-02-28 11:47:45 +00:00
Tim-Philipp Müller
b827e4d539 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
2007-02-07 13:05:01 +00:00
Stefan Kost
66727fe484 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Also mention that a conversion from double to float is suboptimal still.
2007-02-06 14:00:31 +00:00
Tim-Philipp Müller
b7cf10eb6d gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
2007-02-02 11:21:48 +00:00
Stefan Kost
36180a5f3d gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
2007-02-02 09:48:53 +00:00
Wim Taymans
75d5fcb62e gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Don't fail on 0 sized buffers. Fixes #396835.
2007-01-30 11:29:17 +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
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
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
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
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
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
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
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
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
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
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
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
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
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
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