| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
(Fixes #165997)
Original commit message from CVS:
* configure.ac: Put DEFAULT_AUDIOSINK in config.h and use
whereever possible. (Fixes #165997)
* examples/capsfilter/capsfilter1.c: (main):
* examples/dynparams/filter.c: (create_ui):
* examples/seeking/cdparanoia.c: (get_track_info), (main):
* examples/seeking/chained.c: (main):
* examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
(make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
(make_vorbis_pipeline), (make_mp3_pipeline), (make_avi_pipeline),
(make_mpeg_pipeline), (make_mpegnt_pipeline):
* examples/seeking/spider_seek.c: (make_spider_pipeline):
* examples/switch/switcher.c: (main):
* ext/dv/demo-play.c: (main):
* ext/faad/gstfaad.c: (gst_faad_change_state):
* ext/mad/gstmad.c: (gst_mad_chain):
* ext/smoothwave/demo-osssrc.c: (main):
* gst-libs/gst/gconf/gconf.c: (gst_gconf_set_string),
(gst_gconf_render_bin_from_description),
(gst_gconf_get_default_audio_sink),
(gst_gconf_get_default_video_sink),
(gst_gconf_get_default_audio_src),
(gst_gconf_get_default_video_src),
(gst_gconf_get_default_visualization_element):
* gst/level/demo.c: (main):
* gst/level/plot.c: (main):
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/playondemand/demo-mp3.c: (setup_pipeline):
* gst/sine/demo-dparams.c: (main):
* gst/spectrum/demo-osssrc.c: (main):
* gst/speed/demo-mp3.c: (main):
* gst/volume/demo.c: (main):
* testsuite/embed/embed.c: (main):
|
|
Remove float from caps; it doesn't work. Atte...
Original commit message from CVS:
* ext/polyp/polypsink.c: (gst_polypsink_base_init),
(create_context), (gst_polypsink_link): Fix silly endianness
bug. Add some debugging. Remove float from caps; it doesn't
work. Attempt to get remote audio working.
|
|
Original commit message from CVS:
* configure.ac:
* ext/musepack/Makefile.am:
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_class_init),
(gst_musepackdec_init), (gst_musepackdec_dispose),
(gst_musepackdec_src_query), (gst_musepackdec_src_convert),
(gst_musepack_stream_init), (gst_musepackdec_loop),
(gst_musepackdec_change_state):
* ext/musepack/gstmusepackdec.cpp:
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.c: (gst_musepack_reader_peek),
(gst_musepack_reader_read), (gst_musepack_reader_seek),
(gst_musepack_reader_tell), (gst_musepack_reader_get_size),
(gst_musepack_reader_canseek), (gst_musepack_init_reader):
* ext/musepack/gstmusepackreader.cpp:
* ext/musepack/gstmusepackreader.h:
Update to 1.1 API (#165446).
|
|
Original commit message from CVS:
* ext/Makefile.am:
Unbreak buildbot.
|
|
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/amrnb/Makefile.am:
* ext/amrnb/amrnb.c: (plugin_init):
* ext/amrnb/amrnbdec.c: (gst_amrnbdec_get_type),
(gst_amrnbdec_base_init), (gst_amrnbdec_class_init),
(gst_amrnbdec_init), (gst_amrnbdec_link), (gst_amrnbdec_chain),
(gst_amrnbdec_state_change):
* ext/amrnb/amrnbdec.h:
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_get_type),
(gst_amrnbparse_base_init), (gst_amrnbparse_class_init),
(gst_amrnbparse_init), (gst_amrnbparse_formats),
(gst_amrnbparse_querytypes), (gst_amrnbparse_query),
(gst_amrnbparse_handle_event), (gst_amrnbparse_reserve),
(gst_amrnbparse_loop), (gst_amrnbparse_state_change):
* ext/amrnb/amrnbparse.h:
Add support for AMR-NB (mobile phone audio format; #155163, #163286).
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add AMR-NB/-WB raw formats.
* ext/alsa/gstalsa.c: (gst_alsa_link):
Keep valid time when changing format.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_trak):
Add some more format-specific options (#140141, #143555, #155163).
|
|
valid time associated.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_event), (gst_a52dec_chain):
Add some debug output. Check that a discont has a valid
time associated.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
(gst_alsa_sink_loop):
Ignore TAG events. A little extra debug for broken timestamps.
* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
(dvdnavsrc_change_state):
Ensure we send a discont to engage the link before we send any
other events.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
(dvdreadsrc_finalize), (_close), (_open), (_seek_title),
(_seek_chapter), (seek_sector), (dvdreadsrc_get),
(dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
dvd://title works in totem because typefinding sends a seek that ends
up going back to chapter 1 regardless.
* ext/mpeg2dec/gstmpeg2dec.c:
* ext/mpeg2dec/gstmpeg2dec.h:
Output correct timestamps and handle disconts.
* ext/ogg/gstoggdemux.c: (get_relative):
Small guard against a null dereference.
* ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
(gst_textoverlay_set_property):
Free memory when done. Don't call gst_event_filler_get_duration on
EOS events. Use GST_LOG and GST_WARNING instead of g_message and
g_warning.
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
(draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain):
Draw solid lines, prettier colours.
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
Add a default palette that'll work for some movies.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
(gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
(gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
(gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
* gst/mpegstream/gstmpegparse.h:
Use PTM/NAV events when for timestamp adjustment when connected to
dvdnavsrc. Don't use many discont events where one suffices.
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (gst_play_base_bin_add_element):
* gst/playback/gstplaybasebin.h:
Make sure we remove subtitles from the same bin we put them in.
* gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
(gst_subparse_buffer_format_autodetect),
(gst_subparse_change_state):
Fix some memleaks and invalid accesses.
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
(oggskel_type_find), (cmml_type_find), (plugin_init):
Some typefind functions for Annodex v3.0 files
* gst/wavparse/gstwavparse.h:
GstRiffReadClass is the correct parent class.
|
|
timestamps
Original commit message from CVS:
Apply patch from Jeffrey C. Ollie. Fixes rate (now always 8kHz) and
adds timestamps
|
|
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
Argh...
|
|
sincity.mp4 when fixating to six channels i...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
Fix off-by-one bug. Fixes warnings during playback of sincity.mp4
when fixating to six channels in Totem.
|
|
Original commit message from CVS:
Don't return SUCCESS from the state change when the parent call fails
|
|
60000/1001)
Original commit message from CVS:
* ext/dv/gstdvdec.c:
* ext/libfame/gstlibfame.c:
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
replace framerate aproximations by their real value
(24000/1001, 30000/1001, 60000/1001)
Finish fixing bug #164049
|
|
Original commit message from CVS:
* ext/musepack/gstmusepackreader.cpp:
* gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
Some work on tags - still doesn't work in playbin...
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Handle events...
|
|
signal emission and fix caps. Fixes #161667.
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init),
(snapshot_handler), (gst_snapshot_sinkconnect),
(gst_snapshot_chain):
Allocate resources when required, fix recursive signal emission
and fix caps. Fixes #161667.
|
|
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/faac/gstfaac.c: (gst_faac_outputformat_get_type),
(gst_faac_class_init), (gst_faac_init), (gst_faac_srcconnect),
(gst_faac_set_property), (gst_faac_get_property):
* ext/faac/gstfaac.h:
Allow for ADTS output (#153434).
|
|
streams.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chain):
Only increment timestamp if it's valid. Fixes raw AAC streams.
|
|
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
Fix segfault (#161667).
|
|
Original commit message from CVS:
these files should not be here
|
|
Original commit message from CVS:
patch by tim, fix obvious crasher
|
|
Original commit message from CVS:
* configure.ac:
Fix indentation, fix v4l2 plugin detection.
* ext/Makefile.am:
Fix libmms location (Maciej, use diff -u!).
* ext/alsa/gstalsa.c: (gst_alsa_init):
Initialize caps cache to NULL.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Only change state on audiosink if it exists.
|
|
Original commit message from CVS:
Add gstmms
|
|
supported, apart from small warning when setting...
Original commit message from CVS:
First commit. To my knowledge should be in working state, playbin
is supported, apart from small warning when setting URI.
|
|
supported, apart from small warning when setting...
Original commit message from CVS:
First commit. To my knowledge should be in working state, playbin
is supported, apart from small warning when setting URI.
|
|
Original commit message from CVS:
* ext/musepack/gstmusepackdec.cpp:
Fetch error return values. Fixes #161624.
* gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
Really EOS.
|
|
Original commit message from CVS:
* configure.ac: look for musepack headers as musepack/*.h
(fixes #159847)
* ext/musepack/gstmusepackdec.h: use <musepack/*.h>
* ext/musepack/gstmusepackreader.h: same
|
|
a trm id from the MusicBrainz database (#...
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init),
(gst_musicbrainz_init), (gst_musicbrainz_chain),
(gst_musicbrainz_set_property), (gst_musicbrainz_get_property):
* ext/musicbrainz/gsttrm.h:
Add support for using a proxy server when getting a trm id from
the MusicBrainz database (#149613).
|
|
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
|
|
Original commit message from CVS:
merge patch from 158258
|
|
Original commit message from CVS:
* ext/musepack/gstmusepackdec.cpp:
There's also floating point libmusepacks.
|
|
increasing timestamps.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
|
|
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcconnect), (gst_faad_chain):
Oops, remove debug.
|
|
Original commit message from CVS:
2004-11-28 Julien MOUTTE <julien@moutte.net>
* configure.ac:
* ext/Makefile.am:
* ext/directfb/Makefile.am:
* ext/directfb/directfbvideosink.c: (gst_directfbvideosink_create),
(gst_directfbvideosink_get_pixel_format),
(gst_directfbvideosink_get_format_from_fourcc),
(gst_directfbvideosink_fixate), (gst_directfbvideosink_getcaps),
(gst_directfbvideosink_sink_link),
(gst_directfbvideosink_change_state),
(gst_directfbvideosink_chain), (gst_directfbvideosink_buffer_free),
(gst_directfbvideosink_buffer_alloc),
(gst_directfbvideosink_interface_supported),
(gst_directfbvideosink_interface_init),
(gst_directfbvideosink_navigation_send_event),
(gst_directfbvideosink_navigation_init),
(gst_directfbvideosink_set_property),
(gst_directfbvideosink_get_property),
(gst_directfbvideosink_finalize), (gst_directfbvideosink_init),
(gst_directfbvideosink_base_init),
(gst_directfbvideosink_class_init),
(gst_directfbvideosink_get_type), (plugin_init):
* ext/directfb/directfbvideosink.h: Adding a first version of
directfbvideosink.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_init): Initializing some
more.
|
|
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.
|
|
Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
|
|
Original commit message from CVS:
Added the polypaudio sink and hooked it into the build system.
|
|
Original commit message from CVS:
* ext/musepack/gstmusepackreader.cpp:
Workaround for older core.
|
|
error checking.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
This seems to be antique leftover. It needs to pass error
checking.
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_deinitsdl), (gst_sdlvideosink_initsdl),
(gst_sdlvideosink_destroy), (gst_sdlvideosink_create),
(gst_sdlvideosink_sinkconnect), (gst_sdlvideosink_chain):
Fix GstXOverlay implementation (#151059).
|
|
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
|
|
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musepack/gstmusepackdec.cpp:
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.cpp:
* ext/musepack/gstmusepackreader.h:
Add musepack decoder.
* ext/faad/gstfaad.c: (gst_faad_base_init):
Make pad templates static.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(plugin_init):
Add musepack typefinder, make mp3 typefinding work halfway stream,
which doesn't actually work yet because id3demux doesn't implement
_get_length().
|
|
assumptions that dispose is only called once, o...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
(gst_alsa_finalize):
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
(gst_cdaudio_finalize):
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
(cdparanoia_finalize):
* ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
* ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_finalize):
* ext/flac/gstflacdec.c: (gst_flacdec_class_init),
(gst_flacdec_finalize):
* ext/flac/gstflacenc.c: (gst_flacenc_class_init),
(gst_flacenc_finalize):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
(gst_gnomevfssink_finalize):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
(gst_gnomevfssrc_finalize):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
(gst_fameenc_finalize):
* ext/nas/nassink.c: (gst_nassink_class_init),
(gst_nassink_finalize):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
(gst_sdlvideosink_class_init):
* ext/sndfile/gstsf.c: (gst_sf_dispose):
* gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
* gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
* gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_x_window_listener_dispose):
* gst/audioscale/gstaudioscale.c:
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
(play_on_demand_finalize):
* gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
(cdplayer_finalize):
* sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
(gst_glimagesink_class_init):
* sys/oss/gstosselement.c: (gst_osselement_class_init),
(gst_osselement_finalize):
* sys/oss/gstosssink.c: (gst_osssink_dispose):
* sys/oss/gstosssrc.c: (gst_osssrc_dispose):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
Fixes a bunch of problems with finalize and dispose functions,
either assumptions that dispose is only called once, or not calling
the parent class dispose/finalize function
|
|
Original commit message from CVS:
fix build on older automake
|
|
Original commit message from CVS:
* ext/Makefile.am:
Fix the build fixes.
|
|
Original commit message from CVS:
I lied, I am the One True Buildmaster
|
|
Original commit message from CVS:
fix the build
|
|
Original commit message from CVS:
* ext/cdaudio/gstcdaudio.c: (_do_init), (gst_cdaudio_base_init),
(gst_cdaudio_get_event_masks), (gst_cdaudio_send_event),
(gst_cdaudio_query), (plugin_init), (cdaudio_uri_get_type),
(cdaudio_uri_get_protocols), (cdaudio_uri_get_uri),
(cdaudio_uri_set_uri), (cdaudio_uri_handler_init):
Added uri handler for cd://
Port to new API.
|
|
Original commit message from CVS:
* configure.ac: update for swfdec-0.3 and liboil-0.2
* ext/swfdec/gstswfdec.c: update for swfdec-0.3
* ext/swfdec/gstswfdec.h: same
* gst/videofilter/gstvideobalance.c: update for liboil-0.2
* gst/videotestsrc/videotestsrc.c: same
|
|
Original commit message from CVS:
remove unused KIO_DIR
|
|
Original commit message from CVS:
fixing libmng build
|
|
Original commit message from CVS:
commiting patch from Phil Blundell
|
|
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_sinkconnect),
(gst_faad_chain), (gst_faad_change_state):
* ext/faad/gstfaad.h:
Allow playback of raw (unframed) MPEG AAC files (#148993).
|
|
Original commit message from CVS:
remove last mention of kio plugin (in the dist section)
|
