Commit graph

1964 commits

Author SHA1 Message Date
Michael Smith
aec03a4535 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Add notification of current stream. Add ability to configure buffer
sizes.
* gst/playback/gsturidecodebin.c:
Add ability to configure buffer sizes for streaming mode.
Bug #561734.
2008-11-24 20:25:24 +00:00
Jon Trowbridge
0bdeaae59e gst/volume/gstvolume.*: Cleanup volume, define and use default values.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_volume), (gst_volume_set_volume),
(gst_volume_get_volume), (gst_volume_set_mute),
(gst_volume_class_init), (gst_volume_init),
(volume_process_double), (volume_process_float),
(volume_process_int32), (volume_process_int32_clamp),
(volume_process_int24), (volume_process_int24_clamp),
(volume_process_int16), (volume_process_int16_clamp),
(volume_process_int8), (volume_process_int8_clamp), (volume_setup),
(volume_transform_ip), (volume_set_property),
(volume_get_property):
* gst/volume/gstvolume.h:
Cleanup volume, define and use default values.
Recalculate new volume and mute setup before processing. Fixes #561789.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add controller unit test. Patch by: Jonathan Matthew
Fix bogus test that messed with basetransform's internal state.
2008-11-24 12:03:11 +00:00
Wim Taymans
9d0c9fe49b gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
Original commit message from CVS:
* gst/videorate/gstvideorate.c:
Add jpeg and png image media types to the caps. Fixes #561436.
2008-11-22 14:44:26 +00:00
Wim Taymans
096efe2fe9 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain):
Don't post an error when we can't configure the volume but post a
warning instead. Fixes #561780.
2008-11-22 14:31:43 +00:00
Jonathan Rosser
6026260635 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
Original commit message from CVS:
Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a zone plate pattern generator based on BBC R&D Report
1978/23 (yeah *that* 1978).  Try 'videotestsrc pattern=zone-plate
kx2=20 ky2=20 kt=1'.
2008-11-21 20:32:56 +00:00
Sebastian Dröge
d36adc543b gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
2008-11-21 15:45:15 +00:00
Michael Smith
5830b42dc5 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix random fat-fingering making this not compile.
2008-11-21 00:04:48 +00:00
Michael Smith
277e46886c gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
If the top-level type of the stream is plain text, don't try to decode
it, matching behaviour of decodebin.
* gst/playback/gstplaysink.c:
If we fail to generate a text chain (e.g. due to missing optional
plugins), don't crash.
2008-11-20 22:11:38 +00:00
David Schleef
b97e582c57 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add "colorspec" property, specifying whether to generate BT.601
or BT.709 video.  This only affects YCbCr values, not RGB, since
if you're generating a 709 test pattern, presumably you want
709 RGB primaries, not 601.  Also add "smpte75" pattern, which
uses 75% colors instead of 100%, since this is often more useful
for testing (and also follows the SMPTE EG-1 guideline).
2008-11-19 00:24:44 +00:00
Alessandro Decina
f39b66e30d gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
Add a "sink-caps" property to decodebin like it's done for decodebin2.
Fixes #560380.
2008-11-18 18:08:42 +00:00
Jan Schmidt
ca161e799f gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
2008-11-14 21:44:33 +00:00
Mark Nauwelaerts
23f10c5403 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
(plugin_init):
Improve typefinding of ISO JPEG2000 mime types.
2008-11-13 21:11:13 +00:00
Wim Taymans
2773fe8f67 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (deactivate_group):
don't try to unlink the selector sinkpad when we don't have it yet. This
can happen if an error occured before the group was complete.
2008-11-13 17:27:37 +00:00
Wim Taymans
79199d884f gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Catch state change errors and stop from the uridecodebin elements
instead of trying to continue in vain.
2008-11-11 15:52:14 +00:00
Thomas Vander Stichele
9b6f3ad0c8 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
Original commit message from CVS:
* gst/adder/gstadder.c:
Change author string after seeing output of gst-inspector.
2008-11-10 13:55:08 +00:00
Wim Taymans
9045e0428e gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Don't try to do crazy things when we only have a text pad without a
video pad. Fixes #559478.
2008-11-10 10:33:26 +00:00
Wim Taymans
8e07d4ec69 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_init), (volume_setup),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume), (volume_get_property):
* gst/volume/gstvolume.h:
Keep negotiated state in a separate variable.
Protect the volume and mute properties with the object lock.
Protect modifying the transform with the transform lock.
2008-11-05 19:18:25 +00:00
Wim Taymans
d9ae8db094 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
Only convert caps to string when debug is enabled.
2008-11-05 12:20:21 +00:00
Stefan Kost
b9d45d9434 Don't install static libs for plugins. Fixes #550851 for -bad.
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
2008-11-04 12:42:18 +00:00
Sebastian Dröge
35acee194f gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
2008-11-03 08:55:49 +00:00
Sebastian Dröge
9c88e06594 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
2008-11-02 09:19:24 +00:00
Sebastian Dröge
fa613940ef gst/speexresample/: Add missing headers to Makefile.am.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
2008-11-01 19:38:36 +00:00
Sebastian Dröge
35a5ad819d gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
2008-10-30 14:55:43 +00:00
Sebastian Dröge
b2cbf8f91d tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
2008-10-30 14:46:31 +00:00
Sebastian Dröge
4681a87cce gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
2008-10-30 13:44:41 +00:00
Sebastian Dröge
d80b5c4aae Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/arch.h:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(_gcd), (gst_speex_resample_transform_size),
(gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_wrapper.h:
* tests/check/elements/speexresample.c: (setup_speexresample),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance):
Add support for double samples as input and refactor the usage
of the different compilation flavors of the speex resampler.
2008-10-30 12:43:44 +00:00
Stefan Kost
087676f09b gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Return the result of parent_class->event().
2008-10-30 11:43:12 +00:00
Sebastian Dröge
f5b4fa17ff gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
2008-10-29 12:11:20 +00:00
Sebastian Dröge
305b2f3d14 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_get_unit_size),
(gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
(gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
instead of GST_DEBUG, ...
2008-10-28 19:30:33 +00:00
Sebastian Dröge
6009490cef gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
2008-10-28 16:28:45 +00:00
Sebastian Dröge
70348d7327 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (audioresample_fixate_caps):
Fixate the rate to the nearest supported rate instead of
the first one. Fixes bug #549510.
2008-10-28 16:25:00 +00:00
Sebastian Dröge
08489d15ed gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_arm4.h:
* gst/speexresample/fixed_arm5e.h:
* gst/speexresample/fixed_bfin.h:
* gst/speexresample/fixed_debug.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/resample.c: (compute_func), (main), (sinc),
(cubic_coef), (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init_frac), (speex_resampler_process_native),
(speex_resampler_magic), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem):
* gst/speexresample/speex_resampler.h:
Update Speex resampler with latest version from Speex GIT.
2008-10-28 11:46:28 +00:00
Sebastian Dröge
e4e86b0bad gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
Improve MXF typefinding a bit by searching for a header partition
pack instead of just a general partition pack and checking more
bytes for valid values.
2008-10-20 14:08:52 +00:00
Stefan Kost
2cd4c7e2b9 Don't install static libs for plugins. Fixes #550851 for base.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/v4l/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Don't install static libs for plugins. Fixes #550851 for base.
2008-10-16 15:07:00 +00:00
Wim Taymans
5ad1ebcf4c gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
Set the default blocksize to -1 because we will then use the configured
samplesperbuffer to create our output buffer.
2008-10-16 13:50:00 +00:00
Edward Hervey
e68dbb884d gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Don't forget to advance the offset of what we're matching against, else
we end up in a forever loop.
2008-10-15 14:25:50 +00:00
Sebastian Dröge
e86d1dd432 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Improve typefinding a bit. If we don't have a Unicode charset
try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
2008-10-15 11:25:09 +00:00
Sebastian Dröge
e7b42af896 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_convert_to_utf8), (detect_encoding), (convert_encoding),
(get_next_line), (gst_sub_parse_data_format_autodetect),
(feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for UTF16/UTF32 subtitles as long as the first bytes of
the first buffer contain the BOM. This also adds support for other
encodings that allow NUL bytes via the encoding property.
Fixes bugs #552237 and #456788.
2008-10-13 08:58:29 +00:00
Sebastian Dröge
d0d588ff6f gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
For looking at the 4th byte we have to get 4 bytes of course
and not 3.
2008-10-13 08:00:55 +00:00
Sebastian Dröge
862fd1d50f gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Improve FLAC-without-headers typefinding by looking at most of the
frame header and checking if invalid values are used. Should prevent
quite some false positives compared to the old version which only
check if the first 14 bits are set.
2008-10-13 07:52:41 +00:00
Sebastian Dröge
60bf63486b Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
2008-10-10 17:13:40 +00:00
Wim Taymans
81f5117fa9 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_start), (gst_audio_test_src_stop),
(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Define the default property values in the usual place.
Implement start/stop to reset values correctly.
Calculate the sample size only once when we negotiate.
Rename some values to make more sense.
Keep track of our byte range.
Add support for pull based scheduling. Disabled for now until we have
the whole stack working.
Set the BUFFER_OFFSET correctly.
2008-10-10 15:45:15 +00:00
Sebastian Dröge
b735321f58 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
2008-10-10 15:32:10 +00:00
Wim Taymans
fbeec41546 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
Remove bogus assert, the decodepad could have been created inside an
already existing group.
2008-10-08 14:44:04 +00:00
Andy Wingo
3329abde33 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
Original commit message from CVS:
2008-10-08  Andy Wingo  <wingo@pobox.com>

* gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
target instead of setting it.
(gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
API for a decode pad. The bugfix is that we set the group in
activate(), not when the pad was created because it might be NULL
then.
(gst_decode_group_control_source_pad, gst_decode_group_expose):
Update to use the API.
2008-10-08 14:00:07 +00:00
Andy Wingo
6c7e1c8a9b gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
Original commit message from CVS:
2008-10-08  Andy Wingo  <wingo@pobox.com>

* gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
be a subclass of GstGhostPad.
(analyze_new_pad): So, when emitting the signals that determine
how we do autoplugging, already create the ghost pad and use it as
the pad in the signal arguments. This allows applications to make
a connection between the pad passed in e.g. autoplug-continue, and
the pad passed in new-decoded-pad.
(connect_pad, expose_pad): Update to receive the ghosted decode
pad in the args, retargetting it as necessary if we have to plug
the target pad through a multiqueue.
(gst_decode_group_control_source_pad): Adapt to receive an
already-ghosted pad that just needs activation, blocking, and
drain notification.
(sort_end_pads): Adapt for decode pads actually being pads.
(gst_decode_group_expose): Adapt for decode pads actually being
pads. Rewrite the decode pad names so they appear in order. Adds a
new error case if we couldn't set the name.
(gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
logic.
(gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
New API for the decode pad, needed because we shouldn't do these
things inside gst_decode_pad_new(), but after.
(gst_decode_pad_new): Change to actually make the real pad, and
delay the blocking/drainage bits.
2008-10-08 12:49:40 +00:00
Sebastian Dröge
c915582c17 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_convert):
Prevent overflows with big buffer when calculating the size of
the intermediate buffer by using gst_util_uint64_scale() instead of
plain arithmetics. Fixes bug #552801.
2008-10-08 11:50:50 +00:00
Sebastian Dröge
44143f1dcc gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-05 08:11:53 +00:00
Sebastian Dröge
d3edbe7745 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-05 08:10:09 +00:00
Tim-Philipp Müller
029f05635d gst/: Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
2008-09-15 15:11:18 +00:00
Mark Nauwelaerts
ec6afbd321 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Fixes #550638.
2008-09-03 12:23:44 +00:00
Stefan Kost
1875564b65 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
2008-09-03 10:12:04 +00:00
Jonathan Matthew
686a893a0f gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
Original commit message from CVS:
Patch by: Jonathan Matthew  <notverysmart gmail com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for PDF documents (which is nice to have, since it's a
common format, but also helps prevent false positives). Fixes #549814.
2008-08-30 15:55:06 +00:00
Wim Taymans
ed11048c05 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
(no_more_pads_cb):
Fix nasty race where multiple decodebins could start pushing data before
we manage to configure the sinks, resulting in not-linked errors in
typical RTSP streaming cases.
2008-08-27 15:30:16 +00:00
David Schleef
7cce52603e gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: DV typefinding.  Remove
check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Fixes #548065.
2008-08-16 20:57:27 +00:00
Frederic Crozat
89be246154 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-07 15:58:58 +00:00
Andy Wingo
79930b61bf gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
Original commit message from CVS:
2008-08-04  Andy Wingo  <wingo@pobox.com>

* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
documentation fix.
2008-08-04 09:11:08 +00:00
Stefan Kost
7c2a26c9ed gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
Original commit message from CVS:
* gst/adder/gstadder.c:
Cleanup lots of empty lines that came from gst-indent going havoc
before I added the INDENT_ON/OFF marker some time agao.
2008-08-01 13:06:59 +00:00
Wim Taymans
76456cb647 gst/playback/gstplaysink.c: Add some more comments.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
Add some more comments.
2008-07-31 13:06:13 +00:00
Wim Taymans
824a8fc80c gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
(gst_video_test_src_create):
Discard buffers of the wrong size after renegotiation, this is perfectly
possible with things like capsfilter that could suggest caps changes
upstream without knowing the size of the buffer.
2008-07-31 12:58:44 +00:00
Tim-Philipp Müller
58c48279dc gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Dist recently-added gstfastrandom.h.
2008-07-30 19:51:36 +00:00
Stefan Kost
feea3e0b1c gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Fix property doc markup (its not a signal).
* sys/xvimage/xvimagesink.c:
Add since tag for new proeprties (also add sice tags fro the last two
other additions).
2008-07-29 10:26:28 +00:00
Sebastian Dröge
63b89f5625 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
(plugin_init):
Add simple typefinder for the CELT codec (www.celt-codec.org).
2008-07-28 12:47:06 +00:00
Sebastian Dröge
ef5004e56e gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither):
* gst/audioconvert/gstfastrandom.h:
Implement a linear congruential generator as pseudo random number
generator for the dither noise. This is about 2 times faster than
using GLib's mersenne twister. Also this uses only integer math for
generating integers while GLib internally uses floating point math.
2008-07-23 18:34:19 +00:00
David Schleef
cc74285d12 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
Add 'ticks', a 1/30 second sine wave pulse every second.
2008-07-17 02:30:24 +00:00
Jan Schmidt
024d0e56f5 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
First stab at integrating DVD subpicture overlay into
playbin. Successfully plugs and plays, but the queues need
shrinking - 3 seconds of video is too much buffering.
2008-07-14 08:18:58 +00:00
Stefan Kost
8b24a3a057 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Remove now obsolete note in the docs.
2008-07-11 18:06:33 +00:00
Stefan Kost
4f699b7f80 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.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/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-11 06:10:24 +00:00
Stefan Kost
2b33c755b6 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.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/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-10 21:06:06 +00:00
Sebastian Dröge
b02dc1bf6a gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
And ref the pad before returning it again when linking to the queue
failed. Otherwise we will unref the pad twice later and things break.
2008-07-07 09:55:41 +00:00
Sebastian Dröge
ba9c438f98 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
If linking the raw pad with a queue fails, try it without a queue
instead of failing completely. This should never happen.
2008-07-07 09:48:45 +00:00
Evgeniy Stepanov
bddd224b36 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
Original commit message from CVS:
Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
Add a queue after a demuxer if the demuxer outputs raw data. This was
done before only for non-raw data but is required in this case too.
Fixes bug #540215.
decodebin2 doesn't have this issue because all streams of a group
go through multiqueue.
2008-07-06 23:22:12 +00:00
Wim Taymans
3fb8f3b0dd gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_init),
(gst_video_test_src_set_property),
(gst_video_test_src_get_property), (gst_video_test_src_create):
* gst/videotestsrc/gstvideotestsrc.h:
Cleanups, use default property values as defines.
Add property to enable/disable peer buffer allocation.
2008-07-01 13:22:49 +00:00
Sebastian Dröge
a97dc76ad7 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
it on other formats. Also adjust the unit size only for that format
to not include the palette. Fixes bug #540497.
2008-06-30 08:29:09 +00:00
Stefan Kost
e0d27d23cc gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
Original commit message from CVS:
* gst/adder/gstadder.c:
Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
2008-06-29 13:45:27 +00:00
Stefan Kost
69f2aaea3c gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Do not double notify. Remove the unsued return value.
2008-06-24 16:22:45 +00:00
Michael Smith
6ef8ecd7a3 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
Add get-video-pad, get-audio-pad, get-text-pad action signals to
playbin2. This allows the user to get to the selector's sinkpads, and
thus inspect a range of things - caps, tags, etc.
2008-06-20 18:24:24 +00:00
Michael Smith
f5b9e8c065 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Use a different constant for the convert-frame signal id.
Fixes #537009.
2008-06-20 17:27:03 +00:00
Michael Smith
8a59d948d9 gst/playback/: Fix a whole bunch of typos in comments and log statements.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
Fix a whole bunch of typos in comments and log statements.
2008-06-20 17:18:55 +00:00
Michael Smith
9d2564874a gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Ensure decodebin2 emits 'drained' signal once, and only once, when all
pads are drained.
2008-06-20 16:56:18 +00:00
Thomas Vander Stichele
bcc3b3b5f5 apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
Original commit message from CVS:
apparently it's an error to specify nc -l -p 3000 - though the short usage
does not make it very clear that you can drop the host arg with -l
2008-06-20 16:12:50 +00:00
Wim Taymans
cf7da52701 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
(notify_source), (activate_group):
Implement the source property, emit notify when it changes in the
underlying uridecodebin.
2008-06-20 09:19:59 +00:00
Antoine Tremblay
1a71c15677 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
2008-06-20 08:45:13 +00:00
Stefan Kost
332fe99892 Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2008-06-16 07:30:32 +00:00
Michael Smith
2fdd607e95 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.
2008-06-04 17:42:38 +00:00
Tim-Philipp Müller
93db55c074 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
2008-06-04 17:12:40 +00:00
Peter Kjellerstedt
c140528357 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.
2008-06-04 11:33:23 +00:00
Antoine Tremblay
be2f6a8085 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
2008-06-04 05:58:38 +00:00
Sebastian Dröge
d57ab7cfdb gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
2008-06-04 05:44:06 +00:00
Sebastian Dröge
8b14d08115 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
2008-06-04 04:24:27 +00:00
Sebastian Dröge
1d37b272ce gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
2008-06-02 12:20:35 +00:00
Sebastian Dröge
fdd708c418 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
2008-05-30 08:42:17 +00:00
Sebastian Dröge
b86a5d4303 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
2008-05-29 12:17:16 +00:00
Sebastian Dröge
45ef6b5e13 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(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_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously	conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 11:34:09 +00:00
Mark Nauwelaerts
17b17a566f gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.
2008-05-28 14:49:24 +00:00
Sebastian Dröge
57c3aa9b66 gst/adder/gstadder.c: Implement latency query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.
2008-05-28 08:14:47 +00:00
Sebastian Dröge
4ccac97b40 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
2008-05-27 18:10:00 +00:00
Wim Taymans
514b8fa456 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
2008-05-27 10:35:55 +00:00
Tim-Philipp Müller
fa38b99379 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
2008-05-26 10:29:20 +00:00
Tim-Philipp Müller
206f91995b Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
2008-05-25 20:51:35 +00:00
Tim-Philipp Müller
747d52adb3 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.
2008-05-22 22:35:40 +00:00
Jan Schmidt
d58def621b Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
2008-05-22 22:09:16 +00:00
Thijs Vermeir
88b1e8efcf gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.
2008-05-22 18:30:15 +00:00
Sjoerd Simons
1c424d9d93 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.
2008-05-22 11:59:33 +00:00
Julien Moutte
0f80e462d9 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
2008-05-21 16:47:58 +00:00
Wim Taymans
2cdf18edff Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
2008-05-21 16:44:15 +00:00
Wim Taymans
c6b54c3d02 Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
Sebastian Dröge
736b181916 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
2008-05-21 11:36:37 +00:00
Sebastian Dröge
7d605d4514 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
2008-05-21 11:30:58 +00:00
Henrik Eriksson
10ae17ced1 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.
2008-05-21 11:29:25 +00:00
Sebastian Dröge
d47bd6d7bc gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
2008-05-21 07:28:04 +00:00
Antoine Tremblay
a8dda35c1b gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
2008-05-21 06:45:22 +00:00
Sebastian Dröge
e66b0a6642 gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
2008-05-21 05:48:05 +00:00
Sebastian Dröge
fcda3964dc gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
2008-05-20 12:26:32 +00:00
Sebastian Dröge
d76c4b4c65 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
2008-05-20 12:15:34 +00:00
Sebastian Dröge
b5a5d64713 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
2008-05-20 08:12:19 +00:00
Tim-Philipp Müller
7cb1276dac gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
2008-05-19 15:59:40 +00:00
Tim-Philipp Müller
cfc8f3c0d7 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
2008-05-19 14:09:08 +00:00
Sebastian Dröge
05cf63634e gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
2008-05-16 21:12:02 +00:00
Tim-Philipp Müller
d92ff26d29 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:57:41 +00:00
Sebastian Dröge
6720c5beb8 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
2008-05-14 10:58:52 +00:00
Hannes Bistry
b9bc12afd8 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.
2008-05-13 16:02:19 +00:00
Sebastian Dröge
05349cc354 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
2008-05-13 13:04:24 +00:00
Sebastian Dröge
4d5870847f gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.
2008-05-13 09:14:44 +00:00
Tim-Philipp Müller
fed34307db gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
2008-05-10 20:16:21 +00:00
Tim-Philipp Müller
104fed4d66 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
2008-05-10 18:19:17 +00:00
Sebastian Dröge
531c6fb462 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
2008-05-09 08:34:52 +00:00
Edward Hervey
9fa3d7a294 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
2008-05-08 17:35:44 +00:00
Sjoerd Simons
09163ca363 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Sebastian Dröge
b9a285021c gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
2008-05-06 12:35:09 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 12:12:16 +00:00
Wim Taymans
4a3db41f6d gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
2008-05-06 10:16:49 +00:00
Sebastian Dröge
9333eb4899 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 12:33:05 +00:00
Young-Ho Cha
76e3ffb61c gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
2008-05-05 11:14:48 +00:00
Sebastian Dröge
de277a5b2a gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
2008-05-05 10:03:51 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Tim-Philipp Müller
005c1c8636 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
2008-05-03 15:45:23 +00:00
Tim-Philipp Müller
ee90cf1969 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
2008-05-03 15:39:04 +00:00
Tim-Philipp Müller
6de5983831 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
2008-05-03 12:09:16 +00:00
Stefan Kost
2b843ca69f gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
2008-05-02 11:13:05 +00:00
Thijs Vermeir
32c304ad6f Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
2008-05-02 10:54:51 +00:00
Sebastian Dröge
abbce230e2 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
2008-05-02 10:02:05 +00:00
Tim-Philipp Müller
ea0d78e8e5 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
2008-05-01 19:11:42 +00:00
Tim-Philipp Müller
f8977b9e9e gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
2008-04-30 20:54:56 +00:00
Tim-Philipp Müller
5f6db60a4d gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
2008-04-30 14:37:52 +00:00
Michael Smith
802c45b10b gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
2008-04-28 22:18:49 +00:00
Stefan Kost
2b44c294ff gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Remove obsolete streaminfo code and fix a leak. Fixes #529546
2008-04-24 08:19:35 +00:00
Sebastian Dröge
0c73cdcbc8 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
2008-04-19 20:06:59 +00:00
Edward Hervey
9e630c23db gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Don't validate the payload if there isn't any.
Fixes #525915
2008-04-18 14:54:01 +00:00
Stefan Kost
e6528c39fe gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Remove cpp style commented old code.
2008-04-15 19:06:00 +00:00
Stefan Kost
0d3c154ccf gst/playback/gstdecodebin2.c: Fix signal docs.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix signal docs.
2008-04-15 19:02:10 +00:00
Wim Taymans
f0738f6fd3 docs/design/draft-keyframe-force.txt: Fix typo.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
2008-04-11 01:25:01 +00:00
Wim Taymans
e5bdd95038 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
(gst_queue_src_checkgetrange_function):
Include extra buffering stats in the buffering message.
Implement BUFFERING query.
* gst/playback/gsturidecodebin.c: (do_async_start),
(do_async_done), (type_found), (setup_streaming), (setup_source),
(gst_uri_decode_bin_change_state):
Only add decodebin2 when the type is found in streaming mode.
Make uridecodebin async to PAUSED even when we don't have decodebin2
added yet.
2008-04-09 21:40:17 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00
Stefan Kost
b04f8ef354 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update introspection data.
* ext/ogg/gstoggmux.c:
Document oggmux.
* gst/playback/gstdecodebin2.c:
Don't use gtk-doc style comment start for private stuff, but make it
formatted like this for consistency.
2008-04-03 14:58:06 +00:00
Wim Taymans
c98a370f8c gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
(gst_decode_bin_set_property), (gst_decode_bin_get_property),
(analyze_new_pad), (connect_pad), (expose_pad),
(gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
(gst_decode_group_expose), (gst_decode_group_free),
(do_async_start), (do_async_done), (gst_decode_bin_change_state):
Remove fakesink hack, we can now implement this more elegantly.
Added property to bypass typefinding.
Removed underrun callback and demuxer pad probe, we now use the srcpad
probe to expose groups.
API::sink-caps property
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Guard against multiple emissions of the no_more_pads signal, which
happens when we are dealing with chained oggs.
* gst/playback/gsturidecodebin.c: (remove_decoders),
(make_decoder), (type_found), (setup_streaming), (source_new_pad),
(setup_source):
For streams, use our own typefind element and plug our queue after it.
We will need this to determine the type of buffering to use for the
queue soon.
2008-04-03 12:16:04 +00:00
Wim Taymans
8f77a5d928 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_out_rates),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_set_property):
Update the estimated input data when we push out a buffer.
Add some debug info about the temp file.
Only forward src events when we are not using a temp file.
Don't block the duration query, we need to find something better.
Don't leak the temp filename.
2008-04-02 11:08:05 +00:00
Sebastian Dröge
c82eca96a6 configure.ac: Require GLib 2.12 and liboil 0.3.14.
Original commit message from CVS:
* configure.ac:
Require GLib 2.12 and liboil 0.3.14.
* gst/volume/gstvolume.c: (volume_process_double):
Unconditionally use liboil 0.3.14 function.
2008-04-01 14:01:14 +00:00
Michael Smith
670d6cd1e4 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Check the body CRC (if set) when depayloading.
Fixes #522401.
2008-03-27 15:26:38 +00:00
Wim Taymans
03e571d945 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_is_filled):
The queue is never filled when there are no buffers in the queue at all.
Fixes #523993.
2008-03-24 14:08:22 +00:00
Wim Taymans
ad1cbe1edd gst/playback/gstplaybin2.c: Update some docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (free_group), (gst_play_bin_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
(gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_encoding), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb), (perform_eos), (autoplug_select_cb),
(activate_group), (deactivate_group), (setup_next_source),
(save_current_group), (gst_play_bin_change_state):
Update some docs.
Add new locks and conds to protect pipeline creation and group
switching.
Implement the sub-uri property.
Keep track of pending uridecodebin creation and configure the output
pipeline after all streams are configured.
Propagate subtitle encoding to the uridecodebins.
Implement getting the video/audio/visualisation elements.
Use input-selector for stream switching.
If we are asked to do visualisation, prefer to autoplug raw sinks
instead of sinks that accept encoded data.
2008-03-24 12:26:30 +00:00
Wim Taymans
cd1fed3345 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
(gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
(gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
(gst_play_sink_set_volume), (gst_play_sink_get_volume),
(gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
(gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add methods to get audio/video/vis elements.
Add methods to set the font description for the overlay.
Remove properties, we're using this element with its methods only.
Add support for subtitles.
Rearrange the locking a bit to not use the object lock for protecting
the pipeline construction.
Try to use the volume and mute property on the sink when its available.
Implement the mute option with volume when the sink does not have a mute
property.
Only add volume element when the sink has no volume property.
Only do visualisations with raw audio pads.
2008-03-24 12:15:26 +00:00
Wim Taymans
3640ae2d36 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_factories),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
(gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (no_more_pads_full),
(new_decoded_pad_cb), (gen_source_element), (remove_decoders),
(proxy_autoplug_factories_signal), (make_decoder),
(source_new_pad), (setup_source):
Add a readonly source property and notify.
Add new lock for protecting the construction of the pipeline.
Keep track of the decodebins we plugged.
Correctly proxy the autoplug signal so that it actually continues.
Proxy subtitle-encoding to the decodebins.
2008-03-24 11:54:02 +00:00
Wim Taymans
8eb84372bd gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding),
(gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
(deactivate_free_recursive):
Protect caps property with the object lock.
Protect encoding property with the object lock.
Keep list of elements we added that have the subtitle-encoding property.
Distribute the subtitle-encoding to all of the elements when it
changes.
2008-03-24 11:36:08 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
Sebastian Dröge
2034387d4d gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
2008-03-21 16:46:33 +00:00
Sebastian Dröge
88136fc11a gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_create_silence_buffer),
(gst_audio_convert_transform):
Make audioconvert GAP-aware by outputting silence buffers when the
input has the GAP flag set. This is up to 8x faster.
Based on a patch by Stefan Kost. Fixes bug #517813.
2008-03-21 15:58:44 +00:00
Sebastian Dröge
cd5f49f47f gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_double):
Use oil_scalarmultiply_f64_ns() for double processing when it's
available at compile time.
2008-03-21 15:54:54 +00:00
David Schleef
66935a9872 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:  Oops, revert last change
because -base is in freeze.
2008-03-14 18:42:35 +00:00
William M. Brack
bfdcf07674 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
Original commit message from CVS:
Patch by: William M. Brack
* gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
2008-03-14 17:33:09 +00:00
Wim Taymans
4803dd99e5 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Revert change that caused regression until a real fix is found.
Fixes #522203.
2008-03-14 09:54:44 +00:00
Wim Taymans
6c50e0031a gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Update mode property docs, it's deprecated now.
2008-03-07 16:10:51 +00:00
Wim Taymans
8a822e70be gst/: Remove GstPollMode from gstpoll constructor.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create):
* gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
Remove GstPollMode from gstpoll constructor.
2008-03-07 15:48:51 +00:00
Jan Schmidt
43a120bc57 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
Original commit message from CVS:
* gst/Makefile.am:
GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
them twice
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Add new API to the defs
2008-03-03 23:59:45 +00:00
Sebastian Dröge
6f86b8b8a7 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for IMelody files, using audio/x-imelody.
See bug #519516.
2008-03-03 06:22:39 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
José Alburquerque
f61c20f1d2 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst/playback/gstplaybin2.c:
Make the function signature of the _get_*_tags() functions match
the signature of the vfuncs they implement, ie. return a
GstTagList rather than a GstStructure, which is more correct,
even if one is typedef'ed to the other (#518940).
2008-03-02 18:43:15 +00:00
Wim Taymans
472162d1b1 gst/playback/gstplaybin2.c: Enable vis setting.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
Enable vis setting.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gen_vis_chain):
Implement vis switching while playing.
2008-02-29 12:26:48 +00:00
Peter Kjellerstedt
405571a67e gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/tcp/Makefile.am:
* gst/tcp/fdsetstress.c:
* gst/tcp/gstfdset.c:
* gst/tcp/gstfdset.h:
Removed fdset and stress test, they are now known as GstPoll in
core.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
(gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_queue_buffer),
(gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
(gst_multi_fd_sink_stop):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
(gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
(gst_tcp_gdp_read_caps):
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
(gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
(gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
(gst_tcp_client_src_create), (gst_tcp_client_src_start),
(gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
(gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
(gst_tcp_server_src_create), (gst_tcp_server_src_start),
(gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
* gst/tcp/gsttcpserversrc.h:
Port to GstPoll. See #505417.
2008-02-28 10:54:14 +00:00
Sebastian Dröge
929afcbaa1 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Comment smoke typefinder for now. The smokedec plugin needs one
frame per buffer but we have no parser yet, thus it simply crashes
in most situations.
2008-02-25 07:21:33 +00:00
Sebastian Dröge
9327c2a8ec gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for the smoke video codec. Copied from the jpeg plugin.
2008-02-25 06:48:14 +00:00
Sebastian Dröge
49e1c708bb gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mid_type_find),
(plugin_init):
Add midi typefinder, copied from the timidity plugin.
2008-02-25 06:29:09 +00:00
Tomasz Sałaciński
6ab3a0e0c0 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
2008-02-23 09:51:26 +00:00
Stefan Kost
7278c5871c gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
2008-02-21 08:05:10 +00:00
Wim Taymans
dd1282f45a gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (find_property),
(gst_play_sink_find_property), (gen_video_chain),
(gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
Recursively search the sink element for a last-frame property so that we
can also find the property in autovideosink and friends that don't
always proxy the internal sink properties.
2008-02-20 11:52:28 +00:00
Josep Torra Valles
51528422ca gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
Original commit message from CVS:
2008-02-19  Julien Moutte  <julien@fluendo.com>

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

* gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
typefind lookup to fix typefinding on HD clips.
2008-02-19 16:16:55 +00:00
Tim-Philipp Müller
943d4cdc35 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
Original commit message from CVS:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
Fix up copyright (I rewrote the GStreamer-0.10 code for
this from scratch back in the days).
2008-02-19 15:50:37 +00:00
Wim Taymans
81558d6a94 gst/playback/: Add screenshot conversion code from totem.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
(create_element), (gst_play_frame_conv_convert):
* gst/playback/gstscreenshot.h:
Add screenshot conversion code from totem.
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
(gst_play_bin_class_init), (gst_play_bin_convert_frame),
(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
Implement frame property to get a color-unconverted snapshot.
Implement convert-frame action signal to get a converted snapshot image.
Configure connection speed in uridecodebin.
Document some more properties.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_get_last_frame):
* gst/playback/gstplaysink.h:
Use last-buffer property of the video sink to get a video snapshot.
* tests/examples/seek/seek.c: (shot_cb), (main):
Add snapshot button for playbin2 and use the frame property to save the
frame as a png in the current directory.
2008-02-19 15:02:33 +00:00
Josep Torra Valles
58a9fd3622 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
(plugin_init):
Add typefinding support for h264 elementary streams.
Fixes bug #517420.
2008-02-19 11:45:56 +00:00
Stefan Kost
054842ca82 configure.ac: Require CVS of core for new API in collectpads.
Original commit message from CVS:
* configure.ac:
Require CVS of core for new API in collectpads.
* gst/adder/gstadder.c:
Use new API to make adder sparse stream aware.
2008-02-18 13:51:34 +00:00
Wim Taymans
5fc67f8bd3 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb):
Get the object data correct so that we can remove our channels
correctly.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Add option to disable async behaviour in the sinks when possible. This
makes it possible to avoid an audio queue when dealing with
visualisations.
Add option to add a queue for the audio path.
* tests/examples/seek/seek.c: (clear_streams), (update_streams),
(main):
Disable the vis checkbox to match the defaults of playbin2.
Only get the stream info when we need to.
2008-02-18 11:54:15 +00:00
Wim Taymans
5659831526 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Move tee in front of the audio and vis pipelines.
Add queue for audio for now.
Add visualisation support.
* tests/examples/seek/seek.c: (main):
Visualisation is by default disabled.
2008-02-15 18:38:52 +00:00
Wim Taymans
609daaede3 gst/playback/: Add mute property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_audio_chain):
* gst/playback/gstplaysink.h:
Add mute property.
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Make sure we forward the event only once.
* tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
Add and implement the mute button for playbin2.
2008-02-14 18:24:42 +00:00
Tim-Philipp Müller
1d9e1d6a3d gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
* gst/playback/gstplaysink.c: (gen_audio_chain):
Handle case where we can't create the volume element a bit
better (#514307).
2008-02-11 18:31:43 +00:00
Tim-Philipp Müller
cfe66ed251 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Bump rank of jpeg and png typefinders, which will return maximum
probability in the most common cases (thus short-circuiting more
expensive typefinders like the mp3 one for these two quite common
image types).
2008-02-11 13:03:13 +00:00
Zaheer Abbas Merali
b006ba7afe gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Set is_dynamic as True if there are elements with both request
and sometimes src pad templates instead of breaking out when it
finds the first pad template that is a src.
2008-02-09 10:41:36 +00:00
Wim Taymans
c8bb67d0ca gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
Added marshal for streamselector Tags.
* gst/playback/gstplaybasebin.c: (set_active_source):
Streamselector now selects pads based on the pad object instead of its
name.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (get_group), (get_tags),
(gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
(gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
Remove option to mute streams with the current-a/v/t property, we have
this functionality in the flags.
Add signals to notify when the number of A/V/T channels changed.
Add action signals to get tags for the A/V/T streams.
Implement setting the current A/V/T stream.
Rearrange some things to simplify stream selection.
Implement volume.
* gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
(gst_play_sink_get_volume), (gst_play_sink_set_property),
(gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
(activate_vis), (gst_play_sink_reconfigure):
* gst/playback/gstplaysink.h:
Add and implement volume setting methods.
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_finalize),
(gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad):
* gst/playback/gststreamselector.h:
Add pad properties for tags and status of pads.
Keep tags on pads.
Make active pad selection based on pad object instead of name.
2008-02-08 17:47:37 +00:00
Wim Taymans
b5aaf1e1a9 gst/tcp/gstfdset.h: Remove unused field to same some memory.
Original commit message from CVS:
* gst/tcp/gstfdset.h:
Remove unused field to same some memory.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Mark action signals as such.
2008-02-06 15:07:30 +00:00
Wim Taymans
899330d904 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(get_group), (get_n_pads), (gst_play_bin_get_property),
(pad_added_cb), (no_more_pads_cb), (perform_eos),
(autoplug_select_cb), (deactivate_group):
Remove stream-info, we going for something easier.
Refactor getting the current group.
Implement getting the number of audio/video/text streams.
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_get_property),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Add property for number of pads.
* tests/examples/seek/seek.c: (set_scale), (update_flag),
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (msg_async_done),
(msg_state_changed), (main):
Block slider callback when updating the slider position.
Add gui elements for controlling playbin2.
Add callback for async_done that updates position/duration.
2008-02-01 16:44:21 +00:00
David Schleef
5aad3658f8 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c:
Fix valgrind error on 4tap scaling method.
2008-01-14 01:19:34 +00:00
Tim-Philipp Müller
047fb95bad gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Make sure we error out correctly if we can't activate one of
the elements we've added.  Fixes #508138.
2008-01-08 20:48:00 +00:00
Wim Taymans
9c9f60777a gst/playback/gstplay-enum.*: Add enums for configuration flags.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type), (register_gst_play_flags),
(gst_play_flags_get_type):
* gst/playback/gstplay-enum.h:
Add enums for configuration flags.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (gst_play_bin_set_property),
(gst_play_bin_get_property), (no_more_pads_cb),
(autoplug_select_cb), (gst_play_bin_change_state):
Merge mode with flags.
Add more property getters/setters, defaults and docs.
Add properties to get number of audio/video/text streams.
Create sink object in _init so that we can always rely on it being
there.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gen_video_chain), (gen_audio_chain), (gen_vis_chain),
(activate_vis), (gst_play_sink_reconfigure),
(gst_play_sink_set_flags), (gst_play_sink_get_flags),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Use flags to configure the sink pipelines.
Add tee before audio pipeline so that we can use it for visualisations.
Start working on integrating visualisations.
Remove mode, we can do everything with the flags now.
Add method to configue the sink pipeline.
2008-01-07 11:40:04 +00:00
Sebastian Dröge
3ac84ec4ff gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
Original commit message from CVS:
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_choose_func),
(gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
(volume_setup):
* gst/volume/gstvolume.h:
Use GstAudioFilter as base class for the volume element instead of
plain GstBaseTransform.
2008-01-03 20:33:58 +00:00
Thijs Vermeir
b3739a8e7d gst/subparse/gstssaparse.c: combine if's
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
combine if's
2007-12-29 20:55:39 +00:00
Thijs Vermeir
41cc98e287 gst/subparse/gstssaparse.c: remove duplicate log message
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
remove duplicate log message
2007-12-29 19:23:59 +00:00
Wim Taymans
7cb7bffb9e gst/playback/gstplaybin2.c: Code cleanups.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
(autoplug_select_cb), (activate_group), (deactivate_group),
(setup_next_source), (save_current_group),
(gst_play_bin_change_state):
Code cleanups.
Remove next-uri, we can use the uri property just fine.
Fix some crasher.
Unref uridecodebin when switching.
Fix going to READY.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(gen_video_chain), (gen_text_element), (gen_audio_chain),
(gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_set_flags),
(gst_play_sink_get_flags), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add some locking to make things threadsafe.
* gst/playback/test7.c: (about_to_finish_cb):
Fix test.
2007-12-28 09:00:27 +00:00
Tim-Philipp Müller
bd01fd3a57 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
2007-12-22 12:06:47 +00:00
Tim-Philipp Müller
032e064516 gst/playback/gststreamselector.c: Don't leak event.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event):
Don't leak event.
2007-12-21 22:26:47 +00:00
Tim-Philipp Müller
377bde7868 gst/playback/.cvsignore: Ignore more.
Original commit message from CVS:
* gst/playback/.cvsignore:
Ignore more.
2007-12-20 17:13:37 +00:00
Tim-Philipp Müller
85f189aee5 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* gst/playback/gstplaybasebin.c: (set_subtitles_visible),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(setup_sinks), (playbin_set_subtitles_visible):
Make switching off of subtitles work. To avoid all kind of
problems with unlinking of the subtitle input, we just keep
the subtitle inputs linked as they are and tell textoverlay
not to render them. Fixes #373011.
Other subtitle switching issues (esp. when there are both
external and in-stream subtitles) remain. They'll be solved
in playbin2.
2007-12-20 10:41:29 +00:00
Wim Taymans
e56165db4f gst/playback/gststreamselector.c: Init the pad segment too.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_init):
Init the pad segment too.
2007-12-18 16:21:35 +00:00
David Schleef
c5b66243be gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
Add a "blink" pattern.  Turn on the pain.  Apologies.  It's useful
for testing vertical refresh synchronization.
2007-12-18 01:01:23 +00:00
Sebastian Dröge
248742277c Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
Original commit message from CVS:
* configure.ac:
* gst/volume/gstvolume.c: (gst_volume_init):
Use new gst_base_transform_set_gap_aware() function as volume
correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
for this.
2007-12-15 03:40:34 +00:00
Wim Taymans
671d766d8a gst/playback/gstqueue2.c: Use separate timers for input and output rates.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
(reset_rate_timer), (update_in_rates), (update_out_rates),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_chain), (gst_queue_loop):
Use separate timers for input and output rates.
Pause measuring the output rate when we block for more data.
See #503262.
2007-12-14 18:46:12 +00:00
Christian Schaller
9153699f65 update spec file and add two missing files for disting
Original commit message from CVS:
update spec file and add two missing files for disting
2007-12-14 16:23:06 +00:00
Wim Taymans
2da1bb2538 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_chain):
Pause the timer to measure the input rate when we block because the
queue is filled. See #503262.
2007-12-14 09:24:55 +00:00
Wim Taymans
74e5172181 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
Expose the right pad in the right place with the right element.
2007-12-13 12:31:38 +00:00
Wim Taymans
802b38c200 gst/audioconvert/Makefile.am: Also link to libm.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Also link to libm.
2007-12-10 15:21:41 +00:00
Robin Stocker
7bbbf15ad8 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes #502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
2007-12-08 18:38:39 +00:00
Wim Taymans
356971158c gst/playback/gstplay-enum.*: Add missing files.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type):
* gst/playback/gstplay-enum.h:
Add missing files.
2007-12-06 12:08:21 +00:00
Wim Taymans
f2f9bf045b gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
Original commit message from CVS:
* gst/playback/Makefile.am:
Group decodebin2 and uridecodebin into the same plugin so that they
can share the GEnumType.
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
Add signal to sort factories instead of the more awkward autoplug-select
signal.
Modify autoplug_select so that we can try, skip or expose the
autopluggin of an element on a pad.
* gst/playback/gstfactorylists.c: (compare_ranks),
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
(element_filter), (gst_factory_list_get_elements),
(gst_factory_list_debug), (gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Simplify the API, allow getting elements based on mask.
* gst/playback/gstplay-marshal.list:
Add some more marshallers.
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
(autoplug_select_cb), (activate_group):
Add support for managing non-raw sinks by providing a custom element and
sink list to decodebin2.
Try to plug non-raw sinks when decodebin2 using autoplug-select of
decodebin2.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
* gst/playback/gstplaysink.h:
Add support for raw and non-raw sinks.
Add support to force sinks selected by playbin2.
Don't plug raw converters for non-raw sinks.
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
(plugin_init):
Use right accumulators.
Proxy new signal.
2007-12-05 17:11:48 +00:00
Wim Taymans
11bf488b85 gst/playback/: Refactor some common code to filter factories and check caps compat.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
(get_feature_array), (decoders_filter), (sinks_filter),
(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
(gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Refactor some common code to filter factories and check caps compat.
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad),
(find_compatibles):
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_finalize),
(autoplug_factories_cb), (activate_group):
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(proxy_drained_signal):
Add some more debug info and use factor filtering code.
2007-11-30 17:47:15 +00:00
Julien Moutte
0625160416 configure.ac: Add QuickTime Wrapper plug-in.
Original commit message from CVS:
2007-11-26  Julien Moutte  <julien@fluendo.com>

* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
2007-11-26 13:19:46 +00:00
Stefan Kost
1cfef609d0 gst/: Add GAP-flag support.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Add GAP-flag support.
2007-11-26 12:25:55 +00:00
Sebastian Dröge
dacc06a547 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
2007-11-26 08:43:25 +00:00
Sebastian Dröge
155d1b123d gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
2007-11-23 10:21:31 +00:00
Sebastian Dröge
8edd45dbde gst/audioresample/gstaudioresample.c: Implement latency query.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
2007-11-23 10:21:11 +00:00
Sebastian Dröge
816466b67f gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
2007-11-23 10:01:33 +00:00
Sebastian Dröge
f564ebf8cb gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
2007-11-23 08:48:50 +00:00
Sebastian Dröge
d834d1cb48 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
2007-11-21 10:18:56 +00:00
Sebastian Dröge
d832d9bb16 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
2007-11-20 20:23:25 +00:00
Sebastian Dröge
25c4adab31 gst/speexresample/Makefile.am: Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
Add missing file.
2007-11-20 12:56:00 +00:00
Sebastian Dröge
66f2838c8c Add speexresample to the docs and while at that do a make update.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/speexresample/gstspeexresample.h:
Add speexresample to the docs and while at that do a make update.
2007-11-20 07:47:27 +00:00
Sebastian Dröge
3adf5a8875 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
2007-11-20 07:30:30 +00:00
Sebastian Dröge
7fc30c9d28 Add resample element based on the Speex resampling algorithm.
Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.
2007-11-20 07:02:45 +00:00
Stefan Kost
84b35b401a gst/playback/: Fix the build + little README update.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/test7.c:
Fix the build + little README update.
2007-11-17 15:25:15 +00:00
Wim Taymans
b75b5525da gst/playback/: Add playbin2.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
(eos_cb), (about_to_finish_cb), (main):
Add playbin2.
Added gapless playback example.
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
* gst/playback/gstqueue2.c:
* gst/playback/test.c:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(pad_removed_cb):
* gst/playback/gststreaminfo.h:
Change email.
* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (gst_play_bin_handle_message),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
(drained_cb), (unlink_group), (activate_group),
(setup_next_source), (gst_play_bin_change_state),
(gst_play_bin2_plugin_init):
Added raw first version of playbin2. Does chained oggs and gapless
playback fine. No support for raw sinks yet. No visualisations or
subtitles yet.
* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
(gst_play_sink_class_init), (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(post_missing_element_message), (free_chain), (add_chain),
(activate_chain), (gen_video_chain), (gen_text_element),
(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_send_event), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Added Element that abstracts the sinks and their pipelines for playbin2.
2007-11-16 15:44:48 +00:00
Wim Taymans
3f43bfacd2 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
(gst_selector_pad_class_init), (gst_selector_pad_init),
(gst_selector_pad_finalize), (gst_selector_pad_reset),
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_stream_selector_get_type),
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_set_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_is_active_sinkpad),
(gst_stream_selector_activate_sinkpad),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Improve streamselector, make it select and unselect the current pad more
intelligently.
Subclass GstPad for the sinkpads of the selector.
Handle segments more correctly.
Fix caps negotiation.
Implement release_pad.
2007-11-16 15:05:07 +00:00
Wim Taymans
0df5f5b2e6 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_group_check_if_drained), (source_pad_event_probe),
(remove_fakesink):
Add drained signal fired when decodebin finishes decoding the data.
Remove deprecated STATE_DIRTY message.
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
(analyse_source), (proxy_drained_signal), (make_decoder),
(source_new_pad), (value_list_append_structure_list),
(handle_redirect_message), (handle_message):
Proxy the new drained signal.
Handle pad removed from decodebin.
Handle redirect messages by sorting multiple redirections based on the
connection speed.
2007-11-16 12:51:44 +00:00
Stefan Kost
1c2fae7f8b gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Dont leak ghostpad. Fixes #475451.
2007-11-09 15:54:45 +00:00
Wim Taymans
905945738d Update some more docs and comments.
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
2007-11-09 12:21:52 +00:00
Tim-Philipp Müller
750a724841 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes #491722).
2007-11-06 11:09:30 +00:00
Ole André Vadla Ravnås
05a205860d gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value.  Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
2007-11-01 12:51:57 +00:00
Tim-Philipp Müller
5861f366a0 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes #430677.
2007-10-31 17:54:48 +00:00
Tim-Philipp Müller
4c0e44de0f gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Post nice/more useful error message if we don't have a decoder for
the primary type.
2007-10-30 15:54:46 +00:00
Wim Taymans
b55c61c933 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
Be a bit more useful, unblock the pads after we fired the no-more-pads
signal so that we can use the signal to inspect and connect all pads
without having to keep extra state outside of decodebin.
2007-10-30 15:07:58 +00:00
Wim Taymans
b68d48e6bd gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_continue),
(gst_uri_decode_bin_class_init), (no_more_pads_full):
Implement default signal handler so that we return TRUE when nothing is
connected.
2007-10-30 15:00:06 +00:00
Wim Taymans
8c20347774 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (analyze_new_pad):
Move subtitle encoding property to decodebin2 so that it can set the
property value on all elements that it autoplugs and that require it.
Make caps refcounting more consistent in get/set.
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(make_decoder):
Proxy properties and relevant signals from the internal decodebin.
Make properties MT safe.
2007-10-25 17:36:49 +00:00
Wim Taymans
77cef56895 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
Original commit message from CVS:
Inspired by patch of: René Stadler <mail at renestadler dot de>
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
(find_compatibles):
* gst/playback/gstplay-marshal.list:
Remove the autoplug-sort signal and replace it with a binding friendly
autoplug-select signal.
Add an autoplug-factories signal that can be used to generate a list of
factories to try to autoplug.
Add the GstPad to the autoplugging signal args as it might be needed to
make a good factory selection.
Fix up the marshallers for this. Fixes #407282.
2007-10-24 11:07:57 +00:00
Wim Taymans
d33d2be0ed gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
Make the window for a race in typefind and shutting down smaller until
we figure out the right locking here. Avoids #485753 usually.
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
Remove unneeded lock causing a race in typefind and shutting down.
Fixes #485753.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Also remove sinks when going to NULL because we might not complete the
state change to PAUSED, causing the PAUSED->READY state change not to
happen.
2007-10-16 16:48:38 +00:00
Wim Taymans
b6a80a4e42 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
Fix queue negotiation. See #486758.
2007-10-15 11:38:39 +00:00
Wim Taymans
d0897a3528 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (type_found),
(gst_decode_bin_change_state):
Don't disconnect the have_type signal because we never reconnect it
later on. Instead keep a variable to see if we already detected a type.
2007-10-08 17:12:32 +00:00
Wim Taymans
ecb6c19729 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(type_found):
Unlink the signal handler when we found the type, we're not going to do
anything sensible with more type_found signals anyway.
2007-10-08 10:47:26 +00:00
Wim Taymans
818434b664 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(sdp_check_header), (sdp_type_find), (plugin_init):
Add typefind function for application/sdp.
Remove some old dirac typefind code that was ifdeffed out.
2007-10-01 10:22:46 +00:00
Wim Taymans
9f04b80b90 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_push_one):
Fix compilation wrt printf arguments.
2007-09-21 14:37:26 +00:00
Jan Schmidt
d5996e9c37 Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 17:24:55 +00:00
Wim Taymans
d133f1548e gst/playback/gstqueue2.c: Also fix #476514 for queue2.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_flush), (gst_queue_locked_enqueue),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_push_one), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
Also fix #476514 for queue2.
2007-09-17 16:22:17 +00:00
Julien Moutte
87f2e70427 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
Original commit message from CVS:
2007-09-14  Julien MOUTTE  <julien@moutte.net>

* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
typefind for QCP files (RFC #3625)
2007-09-14 10:42:00 +00:00
Josep Torra Valles
1004fb0603 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
Original commit message from CVS:
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c:
Increase upper limit for audio queue a bit; fixes preroll problem
with playbin and decodebin2 when playing a quicktime trailer with
multichannel audio via http (#464666).
2007-09-11 11:29:12 +00:00
Stefan Kost
3df6b8ad42 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Don't leak request pads. Fixes #475395.
2007-09-10 12:05:34 +00:00
Sebastian Dröge
6fa7788c5d Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
2007-09-09 04:08:48 +00:00
Sebastian Dröge
6d7debb0bb gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
2007-09-05 21:20:12 +00:00
Tim-Philipp Müller
12728158b5 gst/playback/gststreaminfo.c: Fix build.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Fix build.
2007-09-05 14:01:25 +00:00
Stefan Kost
53c6315b6b gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Clean up some half-disabled code and comment.
2007-09-05 10:32:09 +00:00
Johan Dahlin
417107b40e gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
Original commit message from CVS:
2007-09-03  Johan Dahlin  <jdahlin@async.com.br>

* gst/typefind/gsttypefindfunctions.c (plugin_init):
Add an audio/x-nsf typefind function for the nsfdec element.
2007-09-04 01:50:55 +00:00
Renato Filho
ac042e8869 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Included "myth://" on stream_uris list for enable buffering to mythtv files
2007-09-03 20:46:38 +00:00
Stefan Kost
d2d03ba2f6 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
The tcp and subparse plugins are under gst, but not totaly free of
dependencies. Handle selection inconfigure.ac, so that they show up
on the final list of what is build and what is not. Maybe they should
better be moved to ext.
2007-08-30 07:29:55 +00:00
Daniel Díaz
b2f2cfc132 Check if libxml provides HTML parser which subparse needs.
Original commit message from CVS:
Patch by: Daniel Díaz  <yosoy@danieldiaz.org>
* configure.ac:
* gst/Makefile.am:
Check if libxml provides HTML parser which subparse needs.
Fixes #451970.
2007-08-30 06:58:46 +00:00
Tim-Philipp Müller
bed6719df7 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
Convert SSA newline codes into actual newline characters (#470766).
2007-08-29 12:16:46 +00:00
Jan Schmidt
973bbf88af gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
We need to set up delayed-linking whenever the caps are non-fixed,
not just when there are multiple types - use gst_pad_is_fixed()
to test.
2007-08-27 11:59:56 +00:00
Jan Schmidt
fc50d2dc64 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
2007-08-24 15:28:33 +00:00
Davyd
bad084b01e gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
Original commit message from CVS:
Based on a patch by: Davyd <davyd at madeley dot id dot au>
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_int32),
(volume_process_int32_clamp), (volume_process_int24),
(volume_process_int24_clamp), (volume_process_int16),
(volume_process_int16_clamp), (volume_process_int8),
(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
* gst/volume/gstvolume.h:
Add support for int32, int24 and int8 to the volume element.
Fixes #445529.
2007-08-23 20:45:45 +00:00
Stefan Kost
a5e777fac3 Original commit message from CVS:
reviewed by: <delete if not using a buddy>
patch by: <delete if not someone else's patch>
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst/typefind/gsttypefindfunctions.c:
* gst/volume/gstvolume.c:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
2007-08-23 08:33:43 +00:00
Stefan Kost
64b4aedf97 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
Original commit message from CVS:
* gst/volume/gstvolume.c:
Enable liboil for float and add more details about problems with
int16.
2007-08-22 11:20:28 +00:00
Wim Taymans
5c59b5a2aa gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
Only post buffering messages when we are a stream.
2007-08-16 11:20:56 +00:00
Michael Smith
1b7a0df57e gst/audiorate/gstaudiorate.c: Debug output fixes.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
2007-08-10 13:55:44 +00:00
Tim-Philipp Müller
2c9bef0180 gst/: Printf format fixes (#465028).
Original commit message from CVS:
* gst/playback/gstqueue2.c:
* gst/videorate/gstvideorate.c:
Printf format fixes (#465028).
2007-08-10 10:08:05 +00:00
Michael Smith
9f9e76bc99 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
2007-08-09 15:44:02 +00:00
Josep Torra Valles
9730f452ee gst/playback/gstplaybasebin.c: Fixes: #465015
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_remove_probe), (queue_threshold_reached):
Patch by: Josep Torra Valles <josep@fluendo.com>
Fixes: #465015
Make sure we remove the check_queues buffer probe from the
correct queue to avoid racily going back to "buffering 99%" when
buffering is actually complete.
Also, fix the spelling of Josep's surname in the ChangeLog.
2007-08-09 12:06:43 +00:00
Josep Torre Valles
382b710277 Add connection-speed property. Fixes #464690.
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* docs/plugins/gst-plugins-base-plugins.args:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (gen_source_element):
Add connection-speed property. Fixes #464690.
2007-08-08 15:05:22 +00:00
Josep Torre Valles
5e5aa7b402 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (queue_threshold_reached),
(gen_source_element), (setup_substreams),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(gst_play_bin_handle_redirect_message):
Move connection-speed property from playbin to playbasebin so that we
can also configure it in source elements that have the connection-speed
property. Fixes #464028.
Add some debug info here and there.
2007-08-07 14:14:54 +00:00
Sebastian Dröge
5310373def gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes #464079.
2007-08-06 16:42:22 +00:00
Sebastian Dröge
6f397125d1 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes #460422.
Also set the default volume to the default value specified in the
GParamSpec.
2007-08-03 19:53:11 +00:00
Jens Granseuer
ef33f2fdc4 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
2007-08-03 19:40:14 +00:00
Dan Williams
ace9335ae3 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
Original commit message from CVS:
Patch by: Dan Williams <dcbw at redhat dot com>
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_get_streaminfo_value_array):
Don't return NULL when querying the stream info value array but instead
return an empty array. Fixes #459204.
2007-07-23 11:18:35 +00:00
Tim-Philipp Müller
2271ec928f gst/playback/gsturidecodebin.c: Init debug category before using it.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
Init debug category before using it.
2007-07-23 10:41:18 +00:00
Wim Taymans
e59c110631 gst/videorate/gstvideorate.c: Use boilerplate.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
(gst_video_rate_query):
Use boilerplate.
Add latency query, might not be perfect yet but already works a lot
better. Fixes #442557.
2007-07-13 18:12:19 +00:00
Jan Schmidt
b6ee0fa3d6 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
Fix the r_mask test for RGBA32 on little-endian.
Fix a stupid typo that would have obviously broken
compilation on big-endian, if anyone was testing.
2007-07-13 15:52:02 +00:00
Wim Taymans
3bac564cc0 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
(paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add alpha to the color struct.
Use a default alpha value of 255 instead of 128.
2007-07-12 15:02:43 +00:00
Wim Taymans
c03d6a8757 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
(setup_source):
Clear the dynamic pads counter when starting a new uri. This makes
reusing playbin work again.
Fixes #454264.
2007-07-12 12:01:20 +00:00
Jan Schmidt
6fa26a44e3 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
Original commit message from CVS:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
(img_get_alpha_info):
Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
of the existing BGRA32 and RGBA32 formats with the alpha at the other
end of the word. Partially fixes #451908
2007-07-06 11:40:45 +00:00
Wim Taymans
d42ca1fd83 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
(gst_adder_request_new_pad):
Make getcaps more robust by not using the proxycaps function. This makes
sure that we don't end up recursively calling getcaps upstream.
See #316248.
2007-07-03 11:52:47 +00:00
Wim Taymans
d4dfef2a0b gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Include math.h to fix compilation.
2007-06-29 17:21:18 +00:00
Jan Schmidt
cae46813ca gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
format, as produced by some dc1394 cameras like the iSight.
See http://www.fourcc.org/yuv.php#IYU1
2007-06-29 14:47:42 +00:00
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
Wim Taymans
8c05f2ebc9 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
Use other metrics as well when estimating the buffer level.
2007-06-28 11:06:56 +00:00
Wim Taymans
aac5185f3e gst/playback/gstplaybasebin.c: Small debug improvement.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.
2007-06-28 10:21:19 +00:00
Wim Taymans
c198d8000c gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
When creating the groups, allow for a 5 second, unlimited buffers
preroll phase after which we expose the group.
When the group is exposed, use a small number of buffers up to a 2
second limit. Also disconnect the overrun signal from multiqueue when we
exposed the group because it is not needed anymore.
2007-06-28 09:46:11 +00:00
Edward Hervey
fa877be84c ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
The chain should be freed if we error out here, else it will leak.
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
(cleanup_decodebin):
Don't forget to *properly* remove the signals, else it will leak.
2007-06-23 14:44:07 +00:00
Wim Taymans
3b2762a5b2 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
When handling a delayed-caps notification case, mark
the group as dynamic so that the nbdynamic count is
incremented and decremented correctly. Fixes: #449156
Patch by: Wim Taymans <wim@fluendo.com>
2007-06-20 11:09:03 +00:00
David Schleef
c4c28a764a gst/playback/gstqueue2.c: Fix compile error from ignored return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Fix compile error from ignored return value.
2007-06-16 03:42:14 +00:00
Michael Smith
6077bc0124 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
Update tmpbuf for all neccesary rows, not just one, as is required
when downscaling.
Fixes #402076.
2007-06-15 15:23:36 +00:00
Edward Hervey
be1f78d2e2 gst/playback/gstqueue2.c: Fix build on MacOSX.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Fix build on MacOSX.
2007-06-13 18:20:57 +00:00
Wim Taymans
2e541b29d4 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
Original commit message from CVS:
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_enqueue):
Fix a division by zero when the max percent is <= 0. Fixes #446572.
also update the buffering status when receiving events. Fixes #446551.
2007-06-12 08:38:06 +00:00
Thiago Sousa Santos
4d83551490 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
(gst_queue_handle_src_query):
Wait for preroll before attempting to forward a duration query upstream.
Fixes #445505.
2007-06-11 11:32:26 +00:00
Wim Taymans
919029d9c5 gst/playback/gstqueue2.c: Fix compilation.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_range):
Fix compilation.
2007-06-07 09:11:27 +00:00
Thiago Sousa Santos
658fbf5039 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_init),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
(gst_queue_src_activate_pull):
Add pull based scheduling and fix some deadlocks. Fixes #444523.
Does not yet completely work because duration queries upstream won't
block yet.
2007-06-06 13:36:26 +00:00
Wim Taymans
1a31080014 Some more fseeko checks.
Original commit message from CVS:
* configure.ac:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Some more fseeko checks.
2007-06-06 09:08:50 +00:00
Sven Arvidsson
0cffe4be7d gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
2007-06-05 21:36:11 +00:00
Michael Smith
6499fcdc2e gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
2007-06-05 17:08:04 +00:00
Wim Taymans
837d4b1bb9 gst/playback/gstqueue2.c: Include stdio to define fseeko.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
Include stdio to define fseeko.
2007-06-05 17:02:13 +00:00
Wim Taymans
d4bb17ab7a gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (no_more_pads_full),
(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
(gst_uri_decode_bin_change_state):
Make sure we name srcpads uniquely even when using different internal
decodebins.
Signal no-more-pads when no more dynamic elements exist.
Remove pads on cleanup.
2007-06-05 16:17:30 +00:00
Thiago Sousa Santos
73e8934af9 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_class_init),
(gst_queue_init), (gst_queue_finalize),
(gst_queue_write_buffer_to_file), (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_is_empty), (gst_queue_is_filled),
(gst_queue_change_state), (gst_queue_set_temp_location),
(gst_queue_set_property):
Add support for filebased buffering. Fixes #441264.
2007-06-05 16:14:23 +00:00
Wim Taymans
3840b5a20f gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
(caps_notify_group_cb), (gst_decode_group_new),
(gst_decode_group_free):
Add support for delayed caps fixation when autoplugging.
Optimize cases where a multiqueue is not needed/wanted, like right after
anything that is not a demuxer.
2007-06-05 16:05:19 +00:00
Wim Taymans
56e2a6b516 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove_flush),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Add support for remuve_flush.
2007-06-05 16:00:33 +00:00
Wim Taymans
5deb6e096d gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun),
(no_more_pads_full):
Stop buffering when the group is commited because the queues filled up.
Fixes #442024.
2007-05-29 13:38:35 +00:00
Jan Schmidt
d9504cf065 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Handle unknown or invalid pads without crashing, as might occur if
a media file like an mp3 is specified as a subtitle file.
Fixes: #410039
2007-05-24 11:15:32 +00:00
Jan Schmidt
c446f911d4 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
(setup_sinks):
Block the subtitle bin output queue before ghosting it and linking,
then unblock after. This avoids spurious not-linked errors caused
by the queue starting up (because it gets linked when it is ghosted).
Fixes: #350299
2007-05-24 10:19:54 +00:00
Wim Taymans
9b188adc27 Small cleanups.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
2007-05-21 10:25:44 +00:00
Stefan Kost
e7c3ddf3fc gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 15:23:43 +00:00
Wim Taymans
a18a10e81f gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
2007-05-17 16:27:32 +00:00
Tim-Philipp Müller
2cd5f527fe gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
2007-05-17 16:11:03 +00:00
Wim Taymans
d33939800d gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
2007-05-17 15:22:44 +00:00
Wim Taymans
fa972968b2 gst/playback/gstqueue2.c: fix build.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
(apply_segment), (apply_buffer), (update_buffering),
(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_filled),
(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
(plugin_init):
fix build.
2007-05-17 13:36:11 +00:00
Wim Taymans
ae69903ca1 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
2007-05-17 11:57:44 +00:00
Michael Smith
ab76fa091a gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
2007-05-17 11:16:14 +00:00
David Schleef
c655a27ab4 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add support for video/x-raw-bayer.
2007-05-15 03:53:11 +00:00
Jan Schmidt
1e2c327792 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Consolidate and re-work our mpeg system stream detection to probe
more packets and produce a higher confidence result. Fixes a
regression caused by lowering the typefind probability last year
- related to bug #397810. Remove the redundant MPEG-1 specific
typefind function, as the new one detects both MPEG-1 & MPEG-2
happily.
Also cleanup the MPEG elementary and MPEG-TS detection functions a
little.
Tested against my media test directory, with some improvements and
no regressions.
2007-05-11 17:33:43 +00:00
Wim Taymans
56f01bc0cb gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
2007-05-10 15:28:13 +00:00
Sébastien Moutte
c88306fe26 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
2007-05-09 21:17:40 +00:00
Stefan Kost
736a5c082f gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
Original commit message from CVS:
* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
gst_adder_change_state):
* gst/adder/gstadder.h (bps, offset, collect_event, segment,
segment_pending, segment_position, segment_rate):
Handle playback-rate on adder.
2007-05-08 19:24:01 +00:00
Stefan Kost
64a9674bd2 gst/: gst/audiotestsrc/gstaudiotestsrc.c
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
2007-05-04 13:10:07 +00:00
Michael Smith
03e4592e41 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If a buffer doesn't have a timestamp, assume it's contiguous with
the previous buffer, and synthesise timestamps appropriately.
2007-05-03 13:16:21 +00:00
Edward Hervey
25d28aae98 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
There is no sensible way to handle incoming buffers which don't have a
valid timestamp. We therefore discard them and wait for the next one.
2007-05-03 10:47:22 +00:00
Tim-Philipp Müller
997621c9b9 gst/playback/: Better error message for text files.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
* gst/playback/gstdecodebin2.c: (plugin_init):
Better error message for text files.
2007-05-01 18:45:36 +00:00
Julien Moutte
d299d1c063 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
Original commit message from CVS:
2007-04-27  Julien MOUTTE  <julien@moutte.net>

* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.
2007-04-27 15:33:46 +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
Dan Williams
37a334ddb7 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
Don't leak incoming buffer if gst_pad_push() returns a
non-OK flow. Fixes #432755.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Unit test for the above by Yours Truly.
2007-04-24 15:00:07 +00:00
Stefan Kost
d24aff28b2 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_sink_event), (gst_adder_collected):
Fix non-flushing segmented seeks, Fixes #340060 for me
2007-04-23 20:04:28 +00:00
Tim-Philipp Müller
97cff37e11 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
2007-04-21 14:14:24 +00:00
Stefan Kost
23a2a0e224 gst/subparse/: Use GST_DISABLE_XML here
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
Use GST_DISABLE_XML here
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_navigation_send_event):
* sys/xvimage/xvimagesink.h:
Include stdlib.h when using atoi.
* tests/check/elements/playbin.c: (playbin_suite):
Use GST_DISABLE_REGISTRY here
2007-04-20 10:42:24 +00:00
Tim-Philipp Müller
1b4546e52f gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
2007-04-17 10:56:37 +00:00
Vincent Torri
0138ad7e09 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.
2007-04-16 22:20:03 +00:00
Thomas Vander Stichele
66a6f3413c gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, buffer_probe_cb, GST_START_TEST):
Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
2007-04-14 12:34:55 +00:00
Thomas Vander Stichele
7f7c429ab1 add debug
Original commit message from CVS:
add debug
2007-04-13 22:10:58 +00:00
Thomas Vander Stichele
9b7fcfdcae debug changes
Original commit message from CVS:
debug changes
2007-04-13 21:09:04 +00:00
Wim Taymans
807258cc03 gst/videorate/gstvideorate.c: Add some debug.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_chain):
Add some debug.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Added check for videorate changing caps handling. Closes #421834.
2007-04-12 15:00:03 +00:00
Thomas Vander Stichele
8a6b8cfb37 log tweaking
Original commit message from CVS:
log tweaking
2007-04-12 10:38:03 +00:00
Wim Taymans
3e455f8a5b gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event):
Make sure we set the IN_CAPS flag correctly.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Get the IN_CAPS flag before we call functions that mess with the flags.
2007-04-12 10:03:22 +00:00
Thomas Vander Stichele
bf82440579 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Only stamp buffers with offset/offset_end right before they get
pushed.  This ensures offset continuity, which was not the case
before as shown by
gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
2007-04-10 20:37:05 +00:00