Commit graph

1819 commits

Author SHA1 Message Date
Sebastian Dröge
c080bfae6d ffmpegcolorspace/videotestsrc: Use v308 instead of V308 2009-02-13 10:10:25 +01:00
Sebastian Dröge
65c322edf2 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
Only conversions from/to are implemented, which
gives (indirect) support for all possible conversions.

Partially fixes bug #571147.
2009-02-12 19:09:40 +01:00
Sebastian Dröge
79d0fff231 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
Partially fixes bug #571147.
2009-02-12 19:09:40 +01:00
Michael Smith
4713bb3abc Revert "Remove pad-removed handlers after setting the decodebins to NULL."
This reverts commit b36d8f3e11.

This brought back some deadlocks. A small leak is better, for now. Need to
figure out a way to fix the leak properly.
2009-02-10 20:38:58 -08:00
Michael Smith
41314315c7 playbin2: Fix segfault on notify after group change.
If our group has been switched, then we get a selector active-pad
notification, we don't need to notify.
2009-02-10 17:20:12 -08:00
Michael Smith
a264efc627 playbin2: Look for volume/mute properties recursively in audio element.
Rather than only checking for volume property on the audio sink
directly, recursively look for it on sinks within it (if it's a bin).
Allows use of sink-as-volume-control where the application has supplied
an audio-sink bin that includes a real audio sink internally.
2009-02-10 17:20:12 -08:00
Sebastian Dröge
5fc20b9ec5 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
Partially fixes bug #571147.
2009-02-10 17:45:59 +01:00
Stefan Kost
f010a38b0d playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
The flags where present but actually not been taken into account.
2009-02-04 13:56:14 +02:00
Stefan Kost
c6ab453eed audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
The comment will ensure that is is marked properly in the docs and the
GParamSpecflag was causing a duplicated initialisation of the same value.
2009-02-04 13:56:13 +02:00
Stefan Kost
b08c0a9003 audioresample: Only pull in liboil if its actualy used.
Liboil still has quite significant startup overhead especialy on embedded
platforms. In audioresample it was only used for the profiling timer.
2009-02-04 10:31:21 +02:00
Stefan Kost
080493ccff typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
Add comments about the flac format. Tighten the check to not allow values that
refer to headers.
2009-02-03 15:28:50 +02:00
Stefan Kost
0ea2afee42 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark. 2009-02-02 15:45:44 +02:00
Wim Taymans
9996aab207 Fix documentation for autoplug-select
fix the documentation strings for the autoplug-select signal.
Fixes #570142.
2009-02-02 12:54:31 +01:00
Michael Smith
b36d8f3e11 Remove pad-removed handlers after setting the decodebins to NULL.
They do needed cleanup; without this we leak selector requestpads.
2009-01-30 18:30:10 -08:00
Michael Smith
61e81ada2c Unref selector request pad even if we no longer have a selector.
During destruction, we won't have a selector any more, but we still need
to unref the pad to avoid leaking it.
2009-01-30 18:30:10 -08:00
Michael Smith
c799f3f77f Unref source in playbin2's finalize method 2009-01-30 18:30:10 -08:00
Michael Smith
b6cbe7e331 Fix more leaks of pads and elements in gstplaysink.
Don't keep extra references to volume and mute elements; we don't need
to do so.
Ensure we unref pads that we have references to, and release request
pads.
2009-01-30 18:30:10 -08:00
Michael Smith
c34f444174 Avoid leaking all playsinks. Fix some internal leaks.
Playsink was holding references to itself. Don't do that, it's not cool.
Also, free all chains in dispose.
2009-01-30 18:30:10 -08:00
Michael Smith
906502b9bb Unref peer request pad after releasing it, since we hold a reference. 2009-01-30 18:30:10 -08:00
Michael Smith
af8d3c51f0 Fix caps leak in playbin2. 2009-01-30 18:30:10 -08:00
Michael Smith
ef1fa84575 Unref active pad from selector when finding active stream. 2009-01-30 18:30:10 -08:00
Michael Smith
f7abf8ed94 Free uris when finalizing playbin2 instance. 2009-01-30 18:30:10 -08:00
Michael Smith
a2b0229058 Unref pads when iterating over them in analyse_source.
Fixes leak of source's srcpad when using uridecodebin.
2009-01-30 18:30:09 -08:00
Jan Schmidt
5337bc03be Fix compilation warning on Forte 2009-01-30 17:58:15 +00:00
Jan Schmidt
6b1e08f277 Don't do void pointer arithmetic. 2009-01-30 17:16:39 +00:00
Michael Smith
81cc88326f Ensure we have sufficient data when using data scan contexts.
Fixes crashes typefinding things that look like they might contain AAC
data (but probably aren't actually AAC).
2009-01-26 18:02:00 -08:00
Sebastian Dröge
5dfcb63252 Rename files and types from speexresample to audioresample
Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
2009-01-23 12:33:41 +01:00
Wim Taymans
cc8b9ae5e8 Add typefind function for gsm
Because core now supports typefindfactories without a typefind function we can
register a factory fo GSM that will --if all else fails-- assume the file is a
GSM file based on the registered extension.
Fixes #566661.
2009-01-23 11:40:26 +01:00
Wim Taymans
8dd3c2d543 Use more performant link function
We can use gst_element_link_pads() instead of the more generic
gst_element_link() function because we know the pads. This saves some cycles
because the more generic function needs to search for possible compatible caps
etc.
2009-01-23 11:37:45 +01:00
Benjamin Gaignard
336e1346e4 Add typefinder for Mobile XMF. Fixes bug #568707. 2009-01-23 10:19:27 +01:00
Jan Schmidt
c42c6d6da0 Fix use-after-unref problem noticed by Josep Torra Valles, and run
gst-indent
2009-01-22 22:09:47 +00:00
Wim Taymans
397c00ac33 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (groups_set_locked_state):
Provide the right arguments to a debug line.
2009-01-13 14:47:19 +00:00
Wim Taymans
1e5f963882 gst/playback/gstplaybin2.c: Fix some comments and docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_uri), (gst_play_bin_set_suburi),
(no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
(activate_group), (deactivate_group), (groups_set_locked_state),
(gst_play_bin_change_state):
Fix some comments and docs.
Post an error message when we fail to link the selector to the sink.
Remove pushing of EOS, this seems unneeded.
Lock the state of deactivated groups so that they don't accidentally
reactivate when the playbin2 state changes.
Reuse uridecodebins.
Unlock and relock state of groups when playbin goes to NULL.
Fixes #566654.
Fixes #566341.
* gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
Only do something in the pad removed callback when we are dealing with
our sourcepads because the sinkpads don't have a ghostpad.
2009-01-07 13:52:14 +00:00
Wim Taymans
8632fc5545 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
Disconnect signal handlers before destroying a previous decodebin so
that we don't end up causing deadlocks. Fixes #566586.
2009-01-05 12:18:52 +00:00
Wim Taymans
c3ec18af97 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_check_get_range),
(gst_audio_test_src_set_property),
(gst_audio_test_src_get_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add property to control pull/push based scheduling.
2009-01-05 10:59:35 +00:00
Wim Taymans
dba6f1f28c gst/playback/gstplaybin2.c: Add some debug info.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Add some debug info.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_reconfigure), (gst_play_sink_request_pad),
(gst_play_sink_release_pad):
Add some more debug info.
Reconfigure the audio chain when we switch between raw and encoded audio
in gapless playback.
2008-12-20 12:48:43 +00:00
Stefan Kost
5a30245c38 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
Original commit message from CVS:
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
Cleanup variable names to make the adder-loop easier to understand.
Also try to use liboil to spee it up, but ifdef it out as it does not
make any change for me (Intel pentim M (sse,sse2) please try on other
systems).
2008-12-17 08:51:34 +00:00
Wim Taymans
0a6d8f01ef Add minimal docs to make the remaining tcp elements show up.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversrc.c:
Add minimal docs to make the remaining tcp elements show up.
Fixes #564139.
2008-12-16 20:16:17 +00:00
Stefan Kost
552b5f5fcd gst/playback/: XRef to GstXOverlay.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
XRef to GstXOverlay.
2008-12-12 13:06:48 +00:00
Edward Hervey
89413e390c gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
Free the factory array when finalizing.
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
Use a GstStaticPadTemplate since the src pad caps are fixed.
2008-12-12 10:54:45 +00:00
Edward Hervey
7a83664099 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
Original commit message from CVS:
* gst/subparse/samiparse.c: (sami_context_push_state),
(sami_context_pop_state), (start_sami_element), (end_sami_element):
Some versions of libxml seem to be very picky as to strict formatting
of the input and never 'close' the final </body> tag.
In order to fix that bad behaviour, we trigger the flushing of
remaining data on both </body> and </sami>.
Fixes #557365
2008-12-11 15:49:12 +00:00
Guillaume Emont
d477a37e7e gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
Original commit message from CVS:
Patch by: Guillaume Emont <guillaume at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinders for MS Word files and OS X .DS_Store files to
prevent them to be recognized as MPEG files. Fixes bug #564098.
2008-12-11 12:32:03 +00:00
Wim Taymans
08736ec1ae gst/playback/gstplaysink.c: Add some more debug info.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain),
(gst_play_sink_reconfigure):
Add some more debug info.
Fix linking of just an encoded sink.
Handle failure to create a sink chain more gracefully than crashing.
2008-12-11 11:04:14 +00:00
Wim Taymans
172e478fd1 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb):
Error out with a missing-plugin error when the input-selector was not
found.
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Indentation.
2008-12-10 18:43:32 +00:00
Wim Taymans
1bfdc87815 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_send_event), (gst_play_sink_change_state):
Use G_DEFINE_TYPE.
Try to set the selected sink to READY before using it. This will allow
for detection of incompatible formats sooner.
Don't cause a fatal error when conversion elements are missing but post
a missing-element message and a warning instead because things might
still link and run fine.
Simplyfy the construction of audio and video sink chains.
2008-12-10 17:39:32 +00:00
Luis Menina
a4493595a6 gst/: Include glib.h instead of a specific GLib header. Including single
Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
2008-12-10 08:19:13 +00:00
Wim Taymans
cf0efcbff9 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_before_transform), (volume_transform_ip):
Use new basetransform vmethod to reconfigure the dynamic properties and
any pending volume/mute changes. Fixes #563508.
2008-12-08 18:44:22 +00:00
Stefan Kost
16e2bccc61 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
Add basic docs to decodebin and link to decodebin from decodebin2.
2008-12-08 15:25:13 +00:00
Christian Schaller
1048c62b2c fix build
Original commit message from CVS:
fix build
2008-11-28 13:30:36 +00:00
Sebastian Dröge
c514656e3a Update documentation of speexresample for the new element name.
Original commit message from CVS:
* 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-videorate.xml:
* gst/speexresample/gstspeexresample.c:
Update documentation of speexresample for the new element name.
2008-11-28 09:44:12 +00:00
Sebastian Dröge
f2eebf3fd3 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
Original commit message from CVS:
* gst/speexresample/README:
Update README with the latest diff between the Speex resampler
and our copy.
2008-11-28 09:04:46 +00:00
Sebastian Dröge
86144acc9a gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (plugin_init):
Update the debug category from speex_resample to audioresample.
2008-11-28 08:37:50 +00:00
Sebastian Dröge
7afac6e23a Remove audioresample files.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
* tests/check/elements/audioresample.c:
Remove audioresample files.
2008-11-27 19:13:59 +00:00
Sebastian Dröge
153406eef5 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* 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:
* gst/speexresample/gstspeexresample.c: (plugin_init):
* gst/speexresample/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(GST_START_TEST), (test_pipeline):
Rename the moved speexresample to audioresample, integrate into the
build system and remove the old audioresample from the build system.
Fixes bug #558124, #385061, #346218, #116051.
2008-11-27 16:57:09 +00:00
Wim Taymans
db785a1f52 gst/playback/gstplaybin2.c: Fix buffer-duration property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Fix buffer-duration property.
2008-11-25 11:00:55 +00:00
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