Age | Commit message (Collapse) | Author | Files | Lines |
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Original commit message from CVS:
* ext/directfb/dfbvideosink.c:
(gst_dfbvideosink_get_format_from_caps):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse), (qtdemux_type_get), (qtdemux_node_dump_foreach),
(qtdemux_dump_hdlr), (qtdemux_dump_dref), (qtdemux_dump_stsd),
(qtdemux_dump_dcom), (qtdemux_parse_trak), (qtdemux_video_caps),
(qtdemux_audio_caps):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_capture_init), (gst_v4l2src_get_size_limits):
Update for GST_FOURCC_FORMAT API change.
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Original commit message from CVS:
* ext/audioresample/gstaudioresample.c:
* ext/polyp/polypsink.c: (gst_polypsink_sink_fixate):
* gst/librfb/gstrfbsrc.c: (gst_rfbsrc_fixate):
* gst/modplug/gstmodplug.cc:
* sys/glsink/glimagesink.c: (gst_glimagesink_fixate):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
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Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Add support for custom genre tags.
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required.
Original commit message from CVS:
* ext/tarkin/wavelet.c:
* ext/tarkin/wavelet.h:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/vbidec/vbidata.c:
* gst/vbidec/vbiscreen.h:
* sys/dxr3/ac3_padder.c:
* sys/dxr3/dxr3audiosink.c:
* sys/dxr3/dxr3spusink.c:
* sys/dxr3/dxr3videosink.c:
* sys/qcam/dark.c:
Don't use gtk-doc markers for normal comments. Fix
gtk-doc formatting where required.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.h:
Remove got_redirect from class structure as well.
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on the bus instead.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(qtdemux_parse_tree):
Remove 'got-redirect' signal and post element message
on the bus instead.
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Original commit message from CVS:
don't put crap in user-visible strings
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Original commit message from CVS:
2005-10-23 Julien MOUTTE <julien@moutte.net>
* gst/tta/gstttaparse.c: (gst_tta_parse_loop): STOPPED->FAILED.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query):
* gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_get_query_types), (gst_tta_parse_query):
API change fix.
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Original commit message from CVS:
2005-10-18 Julien MOUTTE <julien@moutte.net>
* gst/speed/Makefile.am: Fix build of speed.
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other minor things.
Original commit message from CVS:
Fixed Speed - Recovered featured missed since version 1.37, and changed other
minor things.
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Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
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enum definition
Original commit message from CVS:
* examples/indexing/indexmpeg.c: (main):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
(gst_artsdsink_close_audio), (gst_artsdsink_change_state):
* ext/artsd/gstartsdsink.h:
* ext/audiofile/gstafparse.c: (gst_afparse_open_file),
(gst_afparse_close_file):
* ext/audiofile/gstafparse.h:
* ext/audiofile/gstafsink.c: (gst_afsink_open_file),
(gst_afsink_close_file), (gst_afsink_chain),
(gst_afsink_change_state):
* ext/audiofile/gstafsink.h:
* ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
(gst_afsrc_close_file), (gst_afsrc_change_state):
* ext/audiofile/gstafsrc.h:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
* ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackbin.c: (gst_jack_bin_init),
(gst_jack_bin_change_state):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
* ext/nas/nassink.c: (gst_nassink_open_audio),
(gst_nassink_close_audio), (gst_nassink_change_state):
* ext/nas/nassink.h:
* ext/polyp/polypsink.c: (gst_polypsink_init):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
* ext/sdl/sdlvideosink.h:
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
* ext/sndfile/gstsf.c: (gst_sf_set_property),
(gst_sf_change_state), (gst_sf_release_request_pad),
(gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
* ext/sndfile/gstsf.h:
* ext/swfdec/gstswfdec.c: (gst_swfdec_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
* gst/apetag/apedemux.c: (gst_ape_demux_init):
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
* gst/festival/gstfestival.c: (gst_festival_change_state):
* gst/festival/gstfestival.h:
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
* gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
(gst_multifilesink_set_location), (gst_multifilesink_open_file),
(gst_multifilesink_close_file), (gst_multifilesink_next_file),
(gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
(gst_multifilesink_chain), (gst_multifilesink_change_state):
* gst/multifilesink/gstmultifilesink.h:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
(dxr3audiosink_open), (dxr3audiosink_close),
(dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
(dxr3audiosink_change_state):
* sys/dxr3/dxr3audiosink.h:
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
(dxr3spusink_close), (dxr3spusink_chain),
(dxr3spusink_change_state):
* sys/dxr3/dxr3spusink.h:
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
(dxr3videosink_open), (dxr3videosink_close),
(dxr3videosink_write_data), (dxr3videosink_change_state):
* sys/dxr3/dxr3videosink.h:
* sys/glsink/glimagesink.c: (gst_glimagesink_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
(gst_qcamsrc_open), (gst_qcamsrc_close):
* sys/qcam/gstqcamsrc.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
* sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
(gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
(gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
* sys/vcd/vcdsrc.h:
renamed GST_FLAGS macros to GST_OBJECT_FLAGS
moved bitshift from macro to enum definition
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_parse_header):
newsegment API update.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
No need to take stream lock here.
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Original commit message from CVS:
some disting and build fixes
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Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
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Original commit message from CVS:
tta plugin ported to 0.9
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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.
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Original commit message from CVS:
Ported speed Plugin to GStreamer 0.9
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Original commit message from CVS:
Fix up all the state change functions.
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Original commit message from CVS:
cleaning up bad
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Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
fix distcheck
* gst/audioresample/resample.c:
fix wrong docstring
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Original commit message from CVS:
remove stuff
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE.
* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use
the second arg for the class, because G_OBJECT_GET_CLASS (self)
returns the wrong thing.
(gst_signal_processor_add_pad_from_template): Make pads of the
right type.
* ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make
writable param specs G_PARAM_CONSTRUCT so default values work.
(gst_ladspa_init): Use the second arg for the class.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
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Original commit message from CVS:
use base class' newsegment to properly timestamp
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Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
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Original commit message from CVS:
add a check for audioresample
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Original commit message from CVS:
show some info on what's left in the queue
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transform size protect against possible mem corr...
Original commit message from CVS:
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
protect against possible mem corruption and check for discrepancies
between written size and outbuffer's size so we can warn for
potential problems
* gst/audioresample/resample.c: (resample_init),
(resample_get_output_size_for_input), (resample_get_output_size),
(resample_set_n_channels), (resample_set_format):
set debug level based on RESAMPLE_DEBUG env var
make sure that get_output_size* returns a whole number of
sample_size
set sample_size each time either channel or format is set
* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
* gst/audioresample/resample_functable.c:
(resample_scale_functable):
* gst/audioresample/resample_ref.c: (resample_scale_ref):
remove r->sample_size, it's done in resample.c now
add some debugging to the ref implementation
make sure we only give back bytes that are wholes of the sample
size
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Original commit message from CVS:
fix broken header setup in Makefile.am
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Original commit message from CVS:
dist more
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Original commit message from CVS:
port audioresample to basetransform
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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
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Original commit message from CVS:
* configure.ac:
Added mpegaudioparse
* ext/lame/gstlame.c: (gst_lame_src_getcaps),
(gst_lame_src_setcaps), (gst_lame_sink_setcaps),
(gst_lame_sink_event), (gst_lame_chain):
Some cleanups.
Fix memleak.
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_class_init), (gst_mp3parse_init),
(gst_mp3parse_chain), (gst_mp3parse_change_state):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Ported mpegaudioparse
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Original commit message from CVS:
removing README from Makefile.am as its gone from CVS
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Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
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Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event), (gst_faad_chain):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Handle _push() return values.
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Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Fix debug.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps):
Forwardport from 0.8 to implement RLE.
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Original commit message from CVS:
rename
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Original commit message from CVS:
pound some sense in the colorspace elements
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Original commit message from CVS:
licensing, name and description changes
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crash whenever reusing, renegotiating or any...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):
Add debug category, remove Close() call that made it crash
whenever reusing, renegotiating or anything; Close() actually
free()s the handle and should only be called on READY->NULL.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Actually set caps on buffer (in addition to pad), also.
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Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event):
Sign/unsign mismatch.
* configure.ac:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(gst_qtdemux_init), (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
(plugin_init), (gst_qtdemux_handle_sink_event),
(gst_qtdemux_change_state), (gst_qtdemux_loop_header),
(qtdemux_sink_activate), (qtdemux_sink_activate_pull),
(gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
Half-assed port (hey, it works).
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Original commit message from CVS:
2005-07-05 Andy Wingo <wingo@pobox.com>
* Way, way, way too many files:
Remove crack comment from the 2000 era.
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someone will fix it.
Original commit message from CVS:
* configure.ac:
* ext/aalib/Makefile.am:
* ext/aalib/gstaasink.c: (gst_aasink_get_type),
(gst_aasink_class_init), (gst_aasink_fixate), (gst_aasink_setcaps),
(gst_aasink_init), (gst_aasink_get_times), (gst_aasink_render),
(gst_aasink_set_property), (gst_aasink_get_property),
(gst_aasink_open), (gst_aasink_close), (gst_aasink_change_state):
* ext/aalib/gstaasink.h:
* gst/smpte/Makefile.am:
* gst/smpte/gstsmpte.c: (gst_smpte_setcaps), (gst_smpte_init),
(gst_smpte_collected):
* gst/smpte/gstsmpte.h:
Ported 2 more plugins. usgly hack in the Makefile.am though, I'm
sure someone will fix it.
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Original commit message from CVS:
* gst/modplug/gstmodplug.cc:
Add missing break causing position queries to fail.
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position get adjusted when queried (note that c...
Original commit message from CVS:
Add query function to GstSpeed, so that the stream length and current position get adjusted when queried (note that current position queries may still be wrong if the audio sink returns values based on buffer timestamps instead of passing on the query
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