Age | Commit message (Collapse) | Author | Files | Lines |
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DVD, that will cache all data and thus eat...
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
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Prevents player applications from showing...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
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have a demuxer yet.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
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Original commit message from CVS:
2004-08-24 Sebastien Cote <sc5@hermes.usherb.ca>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_element_data), (gst_riff_read_seek),
(gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
#144616, patch reviewed by Ronald and committed by Christophe Fergeau
<teuf@gnome.org>
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Original commit message from CVS:
* gst-libs/gst/video/videosink.h: Change copyright block to LGPL.
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Original commit message from CVS:
* ext/pango/gsttextoverlay.c: Add copyright block and fix plugin
license field
* gst-libs/gst/idct/Makefile.am: Remove mmx/sse code
* gst-libs/gst/video/gstvideosink.c: Change copyright block to
LGPL.
* gst/auparse/gstauparse.c: Fix plugin license field.
* gst/monoscope/gstmonoscope.c: Fix plugin license field.
* gst/mpeg1sys/gstmpeg1systemencode.c: Fix plugin license field.
* gst/rtp/gstrtp.c: Fix plugin license field.
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Original commit message from CVS:
Finished removing GPL'ed MMX code.
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gst-plugins is not supposed to be GPL'ed. This co...
Original commit message from CVS:
Remove GPL'ed mmx32idct.c code and supporting code, since logic in gst-plugins
is not supposed to be GPL'ed. This code provided MMX optimisations, but was
never compiled in since configure never set HAVE_LIBMMX anyway.
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Original commit message from CVS:
* examples/dynparams/filter.c: (ui_control_create):
* examples/gstplay/player.c: (print_tag):
* ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
* ext/gdk_pixbuf/gstgdkanimation.c:
(gst_gdk_animation_iter_may_advance):
* ext/jack/gstjack.c: (gst_jack_request_new_pad):
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
(tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
* ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
* gst-libs/gst/media-info/media-info-test.c: (print_tag):
* gst/sine/demo-dparams.c: (main):
* gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
* testsuite/alsa/formats.c: (create_pipeline):
* testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
fixes for G_DISABLE_ASSERT and friends
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(plugin_init):
require mp3 typefinding to have at least MIN_HEADERS valid headers
add typefinding for AAC adts files
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Original commit message from CVS:
don't install marshal header
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Original commit message from CVS:
don't use stupid colorspace, do use hermes, make macro, mark for translation
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Original commit message from CVS:
no need to link in setup stage
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Original commit message from CVS:
more working plugins
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Original commit message from CVS:
rename GStreamer-0.8.lib to libgstreamer.lib
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Original commit message from CVS:
avoid problems with math.h, fix release dependancy
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for build problems
Original commit message from CVS:
add more plugins to the build
add some definitions needed by plugins
fixes for build problems
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Original commit message from CVS:
more plugins supported under windows
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Original commit message from CVS:
cleaned the makefiles
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Original commit message from CVS:
Copy the files where needed after building, cleaner projects
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Original commit message from CVS:
Clean the local include
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Original commit message from CVS:
ok, that was not very clean
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Original commit message from CVS:
Add the preliminary canvas to build plugins on Win32
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(#148021).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Fix double end-to-native symbol conversion (#148021).
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Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
Make sure we don't create 0 sized subbufers in riff-read.
Signal the no more pads signal after reading the avi header.
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another mime-type for alpha rgb. Currently ...
Original commit message from CVS:
* gst-libs/gst/video/video.h:
Added 32 bits RGBA. Not sure if we should use another mime-type
for alpha rgb. Currently the presence of the alpha_mask property
signals an alpha channel. Ronald?
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_auds):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream):
Set codec_data on caps for avidemuxer.
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media types.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps):
Fix the template caps to include some more media types.
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Original commit message from CVS:
ignore more
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Original commit message from CVS:
don't assert in state change
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Original commit message from CVS:
new method. various debugging
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Original commit message from CVS:
use macro to hash lookup
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Original commit message from CVS:
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_get_length_callback), (gst_play_set_location),
(gst_play_seek_to_time), (gst_play_set_data_src),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization), (gst_play_connect_visualization),
(gst_play_get_sink_element):
- add debugging info
- fix looking up sink elements by iterating over complete caps
- put everything except for source and autoplugger in a complete bin
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#144753)
Original commit message from CVS:
* gst-libs/gst/xoverlay/Makefile.am: xoverlay no longer depends
on X. (bug #144753)
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Original commit message from CVS:
clean up install/dist problem
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Original commit message from CVS:
unbreak install
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Original commit message from CVS:
Add name=source to the wavparse pipeline
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Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/tcp/Makefile.am:
* sys/dxr3/Makefile.am:
don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
tarball.
Also add all *.list files that were missing.
* Makefile.am:
add a distcheck hook to ensure the above doesn't happen again.
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Original commit message from CVS:
2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali.org>
fixed a potential leak with previous commit
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
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Original commit message from CVS:
2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali.org>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Added missing refcount, fixes bug #144425
Cheers Tim for finding the bug
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Original commit message from CVS:
real fix thise time : don't use glib 2.4 specific defines
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Original commit message from CVS:
zaheer :
* gst/tcp/gsttcp.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversrc.c:
- portability fix, to compile on OSX
(fixes #143146)
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
- compilation warnings on OSX
(fixes #143153)
me :
* ext/vorbis/vorbisdec.c : sign warning fixes
* gst-libs/gst/mixer/mixertrack.c : forgoten include
to define newly used G_MAXINT32, bad owen, bad
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Original commit message from CVS:
Added property accessors for mixertrack and mixeroptions.
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Original commit message from CVS:
more readable g_error
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the conversion between internal frequency...
Original commit message from CVS:
* gst-libs/gst/tuner/tunerchannel.h:
- add a freq_multiplicator field to make the conversion
between internal frequency unit and Hz
* sys/v4l/gstv4lelement.c:
* sys/v4l2/gstv4l2element.c:
- change default video device to /dev/video0
* sys/v4l/v4l_calls.c:
* sys/v4l2/v4l2_calls.c:
- we only expose frequency to the user in Hz instead of
bastard v4lX unit (either 62.5kHz or 62.5Hz)
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Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
Initialise b_o_s and e_o_s variables
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add some unusual fourcc's from mplayer avi's
* gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
autoplugging.
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between input/output tracks. Add capture/p...
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init),
(gst_alsa_mixer_build_list), (gst_alsa_mixer_get_volume),
(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
(gst_alsa_mixer_set_record), (gst_alsa_mixer_set_option),
(gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_get_type),
(gst_alsa_mixer_options_class_init), (gst_alsa_mixer_options_init),
(gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
* ext/alsa/gstalsamixertrack.h:
Add enumerations (as GstMixerOptions). Make correct distinction
between input/output tracks. Add capture/playback private flag.
Use flag to decide on whether to set capture or playback volumes
or switches. Use playback and record switches.
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/mixer/mixer-marshal.list:
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init),
(gst_mixer_set_option), (gst_mixer_get_option),
(gst_mixer_mute_toggled), (gst_mixer_record_toggled),
(gst_mixer_volume_changed), (gst_mixer_option_changed):
* gst-libs/gst/mixer/mixer.h:
* gst-libs/gst/mixer/mixeroptions.c: (gst_mixer_options_get_type),
(gst_mixer_options_class_init), (gst_mixer_options_init),
(gst_mixer_options_dispose):
* gst-libs/gst/mixer/mixeroptions.h:
Add GstMixerOptions.
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Rename Audio Mixer to OSS Mixer (similar to Alsa Mixer). Fix
broken device detection on computers with multiple OSS sound
cards.
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Original commit message from CVS:
* gst-libs/gst/resample/private.h:
don't use optimizations that are #if 0'ed
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Original commit message from CVS:
hopefully, fix warnings in asfmux on solaris 10/with forte
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foreign compilers (forte) and gtk-doc (in g...
Original commit message from CVS:
third batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/gst-libs/ this time)
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