Commit graph

4034 commits

Author SHA1 Message Date
Stefan Kost
28b46c1e5d gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-11-01 08:06:13 +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
b0cf257c79 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
2007-10-31 17:32:22 +00:00
Tim-Philipp Müller
55a3eaafea gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 15:30:15 +00:00
Stefan Kost
e37568c196 tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:47:41 +00:00
Tim-Philipp Müller
cfecc1f025 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
2007-10-31 12:30:28 +00:00
Stefan Kost
ffa52e2eac Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
2007-10-30 20:32:14 +00:00
Stefan Kost
19867dfaaf tests/icles/stress-playbin.c: Fix the build.
Original commit message from CVS:
* tests/icles/stress-playbin.c:
Fix the build.
2007-10-30 19:46:02 +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
Sebastian Dröge
a81de45df3 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-28 11:53:36 +00:00
Sebastian Dröge
c3d94683d9 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
Original commit message from CVS:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-28 11:46:48 +00:00
Tim-Philipp Müller
af68c73c6c tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
2007-10-26 18:57:33 +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
Tim-Philipp Müller
698e4050b3 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
Original commit message from CVS:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
2007-10-25 15:10:59 +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
Tim-Philipp Müller
d33e6ca818 gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
2007-10-23 14:23:14 +00:00
Wim Taymans
d3f29fa6e1 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
Make sure that before we clean up the X resources, we shutdown and join
the event thread.
Also make sure the event thread does not shut down immediatly after
startup because the running variable is not yet correctly set.
Fixes #378770.
2007-10-22 10:21:46 +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
6a20747e83 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 15:33:31 +00:00
Tim-Philipp Müller
b7d729d39a tests/icles/: Does what it says on the tin.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/stress-playbin.c:
Does what it says on the tin.
2007-10-16 14:58:53 +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
Jan Schmidt
efb9c3be09 Remove stray character from the changelog.
Original commit message from CVS:
Remove stray character from the changelog.
2007-10-12 10:37:09 +00:00
Jan Schmidt
ce49353432 I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
2007-10-12 10:33:27 +00:00
Tim-Philipp Müller
689ca135f3 Extract vorbis comment LICENSE tags correctly.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Extract vorbis comment LICENSE tags correctly.
2007-10-11 18:24:09 +00:00
Jason Kivlighn
054e43aa3c Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Original commit message from CVS:
Patch by: Jason Kivlighn  <jkivlighn gmail com>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
2007-10-11 16:12:21 +00:00
Tim-Philipp Müller
8458a74526 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
2007-10-10 17:01:51 +00:00
Wim Taymans
02f280a9a0 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 15:36:56 +00:00
Jan Schmidt
c96877779a gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Fix silly search-replace oversight.
2007-10-10 09:37:09 +00:00
Laurent Glayal
0f3c871859 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix caps memleak. Fixes #484989.
2007-10-09 09:57:17 +00:00
Wim Taymans
4f707e2172 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
Fix debug output.
2007-10-08 18:04:34 +00:00
Wim Taymans
c3dda05a8b gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
2007-10-08 18:02:53 +00:00
Jan Schmidt
41ae026041 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
2007-10-08 17:40:17 +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
Tim-Philipp Müller
d3648ca534 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't leak caps.
2007-10-06 16:49:55 +00:00
Tim-Philipp Müller
f277cd8801 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
2007-10-06 15:04:53 +00:00
Sebastian Dröge
6e51063659 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes #483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
2007-10-05 07:49:25 +00:00
Stefan Kost
ff01787e4e gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Replace g_return_if_val (as it could be disabled), with regular return
and warning.
2007-10-04 06:50:53 +00:00
Stefan Kost
fb05ac73ed tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
Original commit message from CVS:
* tests/check/pipelines/simple-launch-lines.c:
Print message name and not just number.
2007-10-03 14:51:59 +00:00
Wim Taymans
5ba1ed3a21 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 11:11:13 +00:00
Stefan Kost
8bef6238c3 tests/examples/snapshot/snapshot.c: Fix the build.
Original commit message from CVS:
* tests/examples/snapshot/snapshot.c:
Fix the build.
2007-10-02 09:04:03 +00:00
Wim Taymans
767366a1e1 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.h:
Add RFC 3556 bandwidth modifiers.
2007-10-01 16:33:00 +00:00
Wim Taymans
d874f65f6e Update documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtppayloads.c:
Update documentation.
2007-10-01 13:37:31 +00:00
Wim Taymans
7cdfb6d154 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
(gst_rtp_payload_info_for_name):
* gst-libs/gst/rtp/gstrtppayloads.h:
Added new file and header to deal with payload info.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Payload specific stuff is move to new headers.
Implement _default_clock rate using the new payload function.
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
(gst_sdp_parse_line):
* gst-libs/gst/sdp/gstsdpmessage.h:
Add some more comments.
2007-10-01 13:22:14 +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
Sébastien Moutte
500d8290a4 win32/common/libgstaudio.def: Add new exported functions.
Original commit message from CVS:
* win32/common/libgstaudio.def:
Add new exported functions.
* win32/vs6/grammar.dsp:
Add autogeneration and copy of some autegenerated files from win32/common
for rtsp library.
* win32/vs6/libgstaudioconvert.dsp:
Add gstaudioquantize.c to the build.
* win32/vs6/libgstinterfaces.dsp:
Add videoorientation.c to the build.
* win32/vs6/libgstriff.dsp:
Add libgsttag to the link libraries list.
* win32/vs6/libgstvolume.dsp:
Add liboil to the link.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstrtsp.dsp:
* win32/common/libgstrtsp.def:
Add files to build libgstrtsp library.
2007-09-29 12:04:02 +00:00
Wim Taymans
724efdb95d tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
Original commit message from CVS:
* tests/examples/snapshot/snapshot.c: (main):
Print error when pipeline failed to construct.
2007-09-26 15:14:37 +00:00
Tim-Philipp Müller
3263d790f9 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Add mappings for the new GST_TAG_COMPOSER for vorbis comments
and ID3v2 tags.
2007-09-25 19:06:47 +00:00
Tim-Philipp Müller
3f7d77fc29 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Don't include config.h in an installed public header, this
might break compilation of applications that don't have such
a header and doesn't necessarily do what it's supposed to do
anyway (ie. check for the lrint/lrintf defines) (#442065).
Add docs for the various macros and document how this header
has to be used (link against libm, etc.); add a few FIXMEs;
include math.h for non-c99 code path.  Based on patch by
Jan Schmidt.
2007-09-25 11:54:09 +00:00
Sebastian Dröge
61988a4b0c configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
Original commit message from CVS:
* configure.ac:
Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
of duplicating these macros in configure.ac.
2007-09-25 07:50:59 +00:00
Thomas Vander Stichele
c550a25a34 po/: Updated translations to 0.10.14
Original commit message from CVS:
* po/hu.po:
* po/sv.po:
* po/uk.po:
Updated translations to 0.10.14
2007-09-22 17:58:22 +00:00
Thomas Vander Stichele
6d6b6b5762 po/pl.po: Added Polish translation.
Original commit message from CVS:
translated by: Jakub Bogusz <qboosh@pld-linux.org>
* po/pl.po:
Added Polish translation.
2007-09-22 17:56:28 +00:00
Thomas Vander Stichele
1f92c12b3e po/fi.po: Added Finnish translation.
Original commit message from CVS:
translated by: Ilkka Tuohela <hile@iki.fi>
* po/fi.po:
Added Finnish translation.
2007-09-22 17:55:37 +00:00
Thomas Vander Stichele
6bd5f909a1 po/es.po: Added Spanish translation.
Original commit message from CVS:
translated by: Jorge González González <aloriel@gmail.com>
* po/es.po:
Added Spanish translation.
2007-09-22 17:54:50 +00:00
Thomas Vander Stichele
53045d6936 po/da.po: Added Danish translation.
Original commit message from CVS:
translated by: Mogens Jaeger <mogens@jaeger.tf>
* po/da.po:
Added Danish translation.
2007-09-22 17:53:59 +00:00
Thomas Vander Stichele
a82eb386af po/zh_CN.po: Added Chinese (simplified) translation.
Original commit message from CVS:
translated by: Funda Wang <fundawang@linux.net.cn>
* po/zh_CN.po:
Added Chinese (simplified) translation.
2007-09-22 17:52:56 +00:00
Thomas Vander Stichele
dcaa63c898 po/bg.po: Added Bulgarian translation.
Original commit message from CVS:
translated by: Alexander Shopov <ash@contact.bg>
* po/bg.po:
Added Bulgarian translation.
2007-09-22 17:51:45 +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
Wim Taymans
476fbd3ddf Add simple snapshot example program using appsink.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/snapshot/.cvsignore:
* tests/examples/snapshot/Makefile.am:
* tests/examples/snapshot/snapshot.c: (main):
Add simple snapshot example program using appsink.
2007-09-20 14:09:24 +00:00
Stefan Kost
2ee06664ce tests/check/generic/states.c: Improved state change unit test.
Original commit message from CVS:
* tests/check/generic/states.c:
Improved state change unit test.
2007-09-20 10:37:02 +00:00
Stefan Kost
ccd019c53b Ignore registries in any format.
Original commit message from CVS:
* docs/plugins/.cvsignore:
* tests/check/.cvsignore:
Ignore registries in any format.
2007-09-19 18:16:58 +00:00
Wim Taymans
2f0a269ddd gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp):
Only copy timestamp on outgoing packets if the depayloader did not set
one.
Also copy duration on outgoing packets.
2007-09-19 16:09:57 +00:00
Wim Taymans
70ad18deee gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix compilation because of missing %d in printf.
When fixating caps, fixate what we can and throw away all remaining
unfixed caps, subclasses should do something smart if they need to.
2007-09-19 15:55:08 +00:00
Stefan Kost
5452f33d1b ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Improve debug logs a bit and be more verbose if things go wrong.
2007-09-19 12:04:21 +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
Wim Taymans
523fd097e6 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Remove code to deal with RTP to GST time conversion, we now just copy
the GST timestamp we receive to the outgoing buffers.
Handle segment and flushes correctly.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
When we have no valid input timestamp, use the previous rtp timestamp on
the outgoing RTP packet instead of the RTP base time.
2007-09-16 19:31:06 +00:00
David Schleef
1beb98e6dc ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
Change alsa alloca's to malloc to fix warnings on gcc-4.2.
2007-09-16 01:56:21 +00:00
Wim Taymans
5dc2aebdee gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
Add some debug info when negotiating caps.
2007-09-15 18:41:27 +00:00
Wim Taymans
a884abc111 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
A buffer with an empty payload is also a valid buffer.
2007-09-15 00:29:11 +00:00
Wim Taymans
06ded6258e gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
(gst_basertppayload_change_state):
Make sure we start our RTP timestamp from the random base RTP
timestamp even if the buffer timestamp starts from some random value.
2007-09-14 20:52:00 +00:00
Wim Taymans
7eb8a5d24a Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/dynamic/.cvsignore:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/addstream.c: (create_stream),
(pause_play_stream), (message_received), (eos_message_received),
(perform_step), (main):
Add simple exmple app to demonstrate starting and pausing live and
non-live bins in a PLAYING pipeline.
2007-09-14 16:56:16 +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
Wim Taymans
4764e6044f gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init):
Disable pull mode scheduling, we're not ready for it yet and it subtly
breaks a lot of things.
2007-09-13 22:52:09 +00:00
Tim-Philipp Müller
998c07972c tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily.  Looks like a problem with the 'jess'
visualisation.
2007-09-12 17:35:52 +00:00
Tim-Philipp Müller
ffdd6b3665 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
2007-09-12 17:15:12 +00:00
Tommi Myöhänen
840c5cd805 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
(read_body), (gst_rtsp_connection_receive):
Make sure we can not cancel in the middle of receiving a message.
Fixes #475731.
2007-09-11 19:07:57 +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
Wim Taymans
c942252430 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Allow othe clocks than the internal clock to be used for the pipeline.
Add property to disable clock provide.
API: GstBaseAudioSrc::provide-clock
2007-09-10 22:10:54 +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
René Stadler
a011ad5aee sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_class_init):
Correctly chain up finalize with the parent class to prevent
memory leaks. Fixes #474880.
2007-09-09 10:25:43 +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
Tim-Philipp Müller
03992b8779 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
2007-09-07 17:37:03 +00:00
Haakon Sporsheim
b2948f2453 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
Original commit message from CVS:
Based on patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
2007-09-07 16:46:05 +00:00
Tim-Philipp Müller
95e3959a5d win32/: Add rtsp enumtypes (#474384) and update others.
Original commit message from CVS:
* win32/MANIFEST:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/gstrtsp-enumtypes.h:
* win32/common/interfaces-enumtypes.c:
* win32/common/interfaces-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
Add rtsp enumtypes (#474384) and update others.
2007-09-07 12:41:01 +00:00
Stefan Kost
7485967d96 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
Original commit message from CVS:
* configure.ac:
Fix configure check for HAVE_LIBXML_HTML.
2007-09-06 20:31:50 +00:00
Tim-Philipp Müller
845e58ecfa tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
Original commit message from CVS:
* tests/check/libs/.cvsignore:
Ignore more, in case the build bots work again one day.
2007-09-06 12:14:25 +00:00
Sebastian Dröge
9022ca9d1b Add libgstfft, a FFT library based on Kiss FFT which is
Original commit message from CVS:
Reviewed by:  Stefan Kost  <ensonic@users.sf.net>
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
(gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
(gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
(gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
(gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
(gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
(gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
(gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
(gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
* gst-libs/gst/fft/gstffts32.h:
* gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
(kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
(kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
(kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
(kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
(kiss_fftr_f32), (kiss_fftri_f32):
* gst-libs/gst/fft/kiss_fftr_f32.h:
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
(kiss_fftr_f64), (kiss_fftri_f64):
* gst-libs/gst/fft/kiss_fftr_f64.h:
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
(kiss_fftr_s16), (kiss_fftri_s16):
* gst-libs/gst/fft/kiss_fftr_s16.h:
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
(kiss_fftr_s32), (kiss_fftri_s32):
* gst-libs/gst/fft/kiss_fftr_s32.h:
* gst-libs/gst/fft/kiss_version:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Add libgstfft, a FFT library based on Kiss FFT which is
BSD licensed. Supported sample formats are int16, int32,
float and double. For those formats a real FFT and IFFT
can be done, different windowing functions can be applied
and functions for extracting the magnitude and phase exist.
Fixes #468619.
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Integrate libgstfft into the docs.
* tests/check/Makefile.am:
* tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
Add unit tests for libgstfft, currently only testing the FFT.
Unit tests for IFFT will follow soon.
2007-09-06 07:00:36 +00:00
Peter Kjellerstedt
f149134834 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(is_multicast_address), (gst_sdp_message_as_text),
(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
(gst_sdp_media_init), (gst_sdp_media_uninit),
(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Separate INIT_ARRAY() and related macros into two versions, one for
structures and one for pointers (e.g., INIT_ARRAY() and
INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
lists of emails and phone numbers.
Add missing const as appropriate.
Change all gint to guint since they all actually represent unsigned
values.
Do not use time as a variable name as it shadows the global time().
Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
Actually implement gst_sdp_message_add_time().
Make gst_sdp_message_add_time() take repeat times as an argument.
Store repeat times in GstSDPTime as a GArray rather than as gchar**.
Corrected the definition of gst_sdp_media_get_bandwidth() (was
misspelled as badwidth).
gst-indented and a little clean up. Fixes #471067.
2007-09-05 23:07:40 +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
7ea0798a9c gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Need to include stdlib.h for abs() here too.
2007-09-05 14:09:15 +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
Wim Taymans
56e39e7c1c gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-04 16:18:48 +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
Wim Taymans
6f93db5ab5 Fix parsing of RB blocks.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix parsing of RB blocks.
Fix docs.
Added helper functions to convert to/from UNIX and NTP time.
API: gst_rtcp_ntp_to_unix()
API: gst_rtcp_unix_to_ntp()
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_get_header_len),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix some more docs.
Implement handling of packets with extensions.
Fix padding check in _validate().
Added function to get extension data.
API: gst_rtp_buffer_get_header_len()
API: gst_rtp_buffer_get_extension_data()
2007-09-03 19:31:11 +00:00
Wim Taymans
0cfb3152b9 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_set_gst_timestamp):
Add some more docs for the queue-delay property and fix a typo in a
comment.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Fix typo.
2007-09-03 19:19:35 +00:00
Wim Taymans
c2460052b3 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
2007-09-03 19:17:33 +00:00
Wim Taymans
210100078d gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Be less silly with the segment start, just apply the clock-base to the
timestamp.
2007-08-31 21:07:20 +00:00
Wim Taymans
827967c8e8 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Deprecate the queue handling thread thing and remove the code.
Use new method to calculate the extended timestamp.
2007-08-31 15:58:30 +00:00
Wim Taymans
27ea51ec37 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_copy_entry):
Use g_strndup which does exactly what we want.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add helper function to compare seqnums.
Add helper function to calculate extended timestamps.
API: gst_rtp_buffer_compare_seqnum()
API: gst_rtp_buffer_ext_timestamp()
2007-08-31 15:21:13 +00:00
Wim Taymans
fdc42d47b4 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_get_entry),
(gst_rtcp_packet_sdes_copy_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix and document SDES item data function.
Add new function that makes a proper copy of SDES item data.
API: gst_rtcp_packet_sdes_copy_entry()
2007-08-30 21:59:23 +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
af6eee1084 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
Original commit message from CVS:
* ext/alsa/gstalsa.c:
Fix typo and compilation on big endian systems.
2007-08-29 14:22:04 +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
Tim-Philipp Müller
b8f1da91d1 API: also add gst_install_plugins_supported() while we're at it (see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
2007-08-28 14:58:17 +00:00
Tim-Philipp Müller
f344ec6b8a API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that.  Fixes #470456.
2007-08-28 14:23:55 +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
Tim-Philipp Müller
e2dbf33a7c gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail):
Add missing separator in PID fallback case.
2007-08-26 14:14:33 +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
Tim-Philipp Müller
f9893ae903 tests/examples/Makefile.am: Fix even more.
Original commit message from CVS:
* tests/examples/Makefile.am:
Fix even more.
2007-08-23 12:37:42 +00:00
Stefan Kost
1772d04dda Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
Original commit message from CVS:
* 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:
* 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:
Revert unwanted commit. many thanks to moap. I want a fix for
https://thomas.apestaart.org/moap/trac/ticket/239
2007-08-23 10:58:42 +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
Wim Taymans
478a6592de gst-libs/gst/audio/audio.c: Clarify the docs a little.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Clarify the docs a little.
2007-08-22 15:29:04 +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
3e3b22148c sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
2007-08-21 15:43:24 +00:00
Wim Taymans
8da7f5ece9 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
When calculating the first timestamp of the buffers, don't go below 0
and clip the samples because the offset was on the eos page.
Fixes #466717.
2007-08-21 12:08:43 +00:00
Wim Taymans
9a32184a05 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
(gst_ogg_demux_collect_chain_info):
Also submit the eos page when trying to find the first timestamp.
See #466717.
2007-08-21 11:42:39 +00:00
Sebastian Dröge
846ddaa550 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 15:24:43 +00:00
Wim Taymans
01d9553d43 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Small cleanups.
On shutdown, don't read the control socket yet.
Set timeout value correctly in all cases.
Add function to check if the server accepts reads or writes.
API: gst_rtsp_connection_poll()
* gst-libs/gst/rtsp/gstrtspdefs.h:
Fix compilation with -pedantic.
Add enum for _poll.
2007-08-17 13:42:49 +00:00
Olivier Crete
b78030f77d gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_getcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add getcaps vfunc to basertppayload. See #465146.
2007-08-16 16:06:21 +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
Tim-Philipp Müller
0afe67c9e0 gst-libs/gst/pbutils/: Small docs fix and addition.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.c:
Small docs fix and addition.
2007-08-15 17:05:45 +00:00
Tim-Philipp Müller
5ff55c7a30 tests/icles/: Add a dumb little test for textoverlay alignments.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-textoverlay.c:
Add a dumb little test for textoverlay alignments.
2007-08-13 15:37:29 +00:00
Dan Williams
4200d788bc ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
Original commit message from CVS:
Patch by: Dan Williams  <dcbw redhat com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
API: add "line-alignment" property (#459334). Add gtk-doc blurb for
"silent" property so there's a Since tag in the API reference.
2007-08-13 15:26:54 +00:00
Thomas Vander Stichele
4147da026a fix ... by: lines
Original commit message from CVS:
fix ... by: lines
2007-08-13 11:21:00 +00:00
Wim Taymans
3b7071a16f gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Improve caps negotiation so that downstream elements can confiure
certain RTP properties by fixing them on the caps. See #465146.
Add docs.
2007-08-12 16:30:36 +00:00
Tim-Philipp Müller
2d5d5ac891 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Mark as deprecated some macros which were presumably meant to be
private API and accidentally exposed in the public header file.
Also actually _init() lock (only works at the moment because the
struct is zeroed out when created and the initial values in the
mutex struct are zeroes too). (#459585)
2007-08-11 12:39:51 +00:00
Stefan Kost
3ad40bebe5 docs/libs/Makefile.am: Remove cruft and do some cleanups.
Original commit message from CVS:
* docs/libs/Makefile.am:
Remove cruft and do some cleanups.
* docs/libs/gst-plugins-base-libs-docs.sgml:
Prepare for comming gtkdoc features (rebase against online docs).
2007-08-10 17:35:52 +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
Stefan Kost
87d96c656a ext/ogg/gstoggmux.c: Do not leak oggmux instance.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Do not leak oggmux instance.
* ext/vorbis/vorbisenc.c:
Also log values.
2007-08-09 11:37:28 +00:00
Thomas Vander Stichele
f6be63b93b po/: Updated translations.
Original commit message from CVS:
* po/hu.po:
* po/it.po:
* po/nl.po:
* po/uk.po:
* po/vi.po:
Updated translations.
2007-08-09 10:51:55 +00:00
Yang Hong
afd8b931a9 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
Original commit message from CVS:
patch by: Yang Hong <hongyang@redflag-linux.com>
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
Add 'silent' property to GstTimeOverlay. Fixes #462979
2007-08-08 16:07:21 +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
Damien Lespiau
9b8c837165 Fix compilation on windows. Fixes #464320.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
Fix compilation on windows. Fixes #464320.
2007-08-07 15:13:46 +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
Wim Taymans
607fa48ad8 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
Add rdt manager for rdt transport.
Fix parsing of RDT transport.
2007-08-03 15:44:01 +00:00
Jan Schmidt
2f6e0e7b57 configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
2007-08-03 14:43:15 +00:00
Jan Schmidt
221ae4ebd7 Release 0.10.14
Original commit message from CVS:
Release 0.10.14
2007-08-03 14:41:46 +00:00
Jan Schmidt
a9f63daff7 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
Original commit message from CVS:
* tests/check/libs/audio.c: (GST_START_TEST):
Fix the test to reflect the behaviour of gst_audio_clip_buffer.
2007-07-27 17:37:19 +00:00
Jan Schmidt
d5dc054ea3 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
When clipping a buffer with no timestamp, assume it is
within the segment without warnings.
Fixes: #460978
2007-07-27 17:10:47 +00:00
Wim Taymans
be5ef4b0ad gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
Fire the signal on the object, not the interface.
2007-07-27 11:16:23 +00:00
Jan Schmidt
1846b1a84d gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ber. Don't include the full path, idiot.
2007-07-27 09:17:19 +00:00
Jan Schmidt
c339ca80c3 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ignore generated files.
2007-07-27 08:29:29 +00:00
Jan Schmidt
aa14635c47 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
* gst-libs/gst/interfaces/rtspextension.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp.h:
* gst-libs/gst/rtsp/gstrtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/rtsp/gstrtspextension.h:
* gst-libs/gst/rtsp/rtsp-marshal.list:
Move the rtspextension.h interface into gstrtspextension.h
as part of libgstrtsp instead of libgstinterfaces, because it's
only for use within plugins, not applications.
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
is abstract.
2007-07-26 19:57:15 +00:00
Wim Taymans
6d1a34eff2 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_iface_init),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Fix marshaller for the send signal.
Add URL to stream selection interface method.
2007-07-26 15:48:01 +00:00
Jan Schmidt
50a3a239a0 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
Pull in our dependencies from -base before those from outside.
2007-07-26 15:35:43 +00:00
Wim Taymans
2c35823bdf API: gst_rtsp_base64_decode_ip()
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
* gst-libs/gst/rtsp/gstrtspbase64.h:
API: gst_rtsp_base64_decode_ip()
Added function to decode Base64 in-place.
2007-07-26 14:33:01 +00:00
Jan Schmidt
58afe32d55 tests/check/libs/.cvsignore: Ignore the mixer test binary.
Original commit message from CVS:
* tests/check/libs/.cvsignore:
Ignore the mixer test binary.
2007-07-26 14:08:01 +00:00
Jan Schmidt
b947924e28 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
Gratuitous comment change to trigger a rebuild on the buildbots.
2007-07-26 10:00:37 +00:00
Wim Taymans
8db50d49f7 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_get_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val):
* gst-libs/gst/sdp/gstsdpmessage.h:
Constify args where we can.
2007-07-25 18:20:36 +00:00
Wim Taymans
256d005e49 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Move interface for RTSP extensions from -good to here.
Added helper methods to invoke interface methods.
2007-07-25 18:18:49 +00:00
Wim Taymans
77c284a31f Fix some more RTSP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
(gst_rtsp_message_init_response),
(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_get_body), (dump_key_value):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
Fix some more RTSP docs.
Add some missing methods for dealing with messages.
2007-07-25 11:22:30 +00:00
Wim Taymans
3dff14d6b1 Added beginnings of RTSP documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (add_auth_header),
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
(read_body), (gst_rtsp_connection_receive),
(gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtspurl.h:
Added beginnings of RTSP documentation.
2007-07-24 19:19:33 +00:00
Wim Taymans
ee42361c89 Document the SDP library.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_media_new),
(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_media_get_media), (gst_sdp_media_set_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_add_format),
(gst_sdp_media_get_information), (gst_sdp_media_set_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_set_key), (gst_sdp_media_get_key),
(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
(print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Document the SDP library.
Add some of the missing SDPMedia methods.
2007-07-24 17:37:03 +00:00
Wim Taymans
19e0dd3140 Move SDP and RTSP from helper objects in -good to a reusable library.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
(parse_response_status), (parse_request_line), (parse_line),
(gst_rtsp_connection_read), (read_body),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
(gst_rtsp_find_method):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_new), (gst_rtsp_message_init),
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
(gst_rtsp_message_dump):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse), (gst_rtsp_range_free):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
(range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
(gst_rtsp_transport_free):
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
(gst_sdp_attribute_init), (gst_sdp_message_new),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
(gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
(read_string), (read_string_del), (gst_sdp_parse_line),
(gst_sdp_message_parse_buffer), (print_media),
(gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Move SDP and RTSP from helper objects in -good to a reusable library.
Use a proper gst_ namespace.
2007-07-24 11:52:56 +00:00
Sebastian Dröge
9137e98926 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
(vorbis_dec_flush_decode):
Use the new buffer clipping function from gstaudio here.
2007-07-23 18:42:22 +00:00
Sebastian Dröge
6be2524031 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 18:26:09 +00:00
Stefan Kost
14e301026d docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Cleanup the docs.
2007-07-23 14:45:16 +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
Jan Schmidt
0776d87e32 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Add padding vars in place of the signal pointers
when building with DISABLE_DEPRECATED so that the
interface structure doesn't change size.
2007-07-21 09:56:09 +00:00
Marc-Andre Lureau
c161e29307 Fixes: #152864
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixertrack.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.c:
* gst-libs/gst/interfaces/mixertrack.h:
* tests/check/Makefile.am:
* tests/check/libs/mixer.c:
Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
Fixes: #152864
Add support for notifying mixer changes on the message bus, and
implement it in alsamixer.
API: gst_mixer_get_mixer_flags
API: gst_mixer_message_parse_mute_toggled
API: gst_mixer_message_parse_record_toggled
API: gst_mixer_message_parse_volume_changed
API: gst_mixer_message_parse_option_changed
API: GstMixerMessageType
API: GstMixerFlags
2007-07-21 09:21:12 +00:00
Michael Smith
11cf0dcd6b sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
xcontext->im_format is only for testing XShm support (as the header
file comments document). Use xvimage->im_format for everything else.
Avoids spurious warnings on buffer allocation before setcaps.
2007-07-20 16:09:03 +00:00
Stefan Kost
9b2fb4d824 tests/: We should use $(LIBM).
Original commit message from CVS:
* tests/examples/volume/Makefile.am:
* tests/icles/Makefile.am:
We should use $(LIBM).
2007-07-20 07:22:15 +00:00
Stefan Kost
f50d9ab580 tests/icles/Makefile.am: This needs -lm.
Original commit message from CVS:
* tests/icles/Makefile.am:
This needs -lm.
2007-07-20 06:13:21 +00:00
Wim Taymans
d0e9a76a95 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property):
Don't break ABI, restore previous ranges. Keep the default random
selection of timestamp and seqnum offset but as soon as the app sets a
specific value, use that one.
2007-07-16 10:10:28 +00:00
Bastien Nocera
312c0bd5ba sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess dot net>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
Add option to turn off double-buffering for debugging purposes.
Fixes #437169.
2007-07-14 18:33:15 +00:00
Jorn Baayen
877fa6035a sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
Original commit message from CVS:
Patch by: Jorn Baayen <jorn at openedhand dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
(gst_ximagesink_set_property), (gst_ximagesink_get_property),
(gst_ximagesink_init), (gst_ximagesink_class_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
add 'handle-expose' property. Useful for video widgets which may want to
be in control of Expose behaviour. Fixes #380625
2007-07-14 18:20:41 +00:00
Wim Taymans
c82275a51d gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Fix ranges of rtp payloader properties so that the full range can be
used in addition to -1 (random).
Fix wrong seqnum reporting in caps.
Fixes #420326.
2007-07-14 17:23:42 +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
476361497d sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_setcaps):
* sys/xvimage/xvimagesink.h:
After a caps change, redraw our borders to avoid garbage left there
when the image format changes to a smaller size, like 16:9 -> 4:3
Also, hold the flow_lock a bit longer in the set_caps while we're
fiddling with the xcontext.
2007-07-13 16:05:17 +00:00
Jan Schmidt
2b8d07bac0 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* tests/Makefile.am:
Remove bogus check for libcheck, since we check for
gstreamer-check and it pulls in the required info from there, and we
weren't actually _using_ the information for libcheck ourselves
anyway.
2007-07-13 16:02:23 +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
Stefan Kost
b5c2a72a4a configure.ac: Use pkg-config to locate check.
Original commit message from CVS:
* configure.ac:
Use pkg-config to locate check.
2007-07-12 11:13:32 +00:00
Tim-Philipp Müller
79b4be47a8 Fix 'make check' build against core CVS.
Original commit message from CVS:
* configure.ac:
* tests/check/elements/volume.c: (GST_START_TEST):
Fix 'make check' build against core CVS.
2007-07-11 23:12:12 +00:00
Stefan Kost
aac0353ce6 gst-libs/gst/: Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
2007-07-10 20:46:41 +00:00
Tim-Philipp Müller
8a499651b9 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
2007-07-08 13:07:38 +00:00
Stefan Kost
e59bb29af4 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Fix location of includes in the docs.
2007-07-06 18:19:39 +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
Stefan Kost
d03f78d47f docs/: Simplify --extra-dir as gtkdoc scans recursively.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Simplify --extra-dir as gtkdoc scans recursively.
2007-07-05 08:43:30 +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
Tim-Philipp Müller
28ef3f5ddf gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
2007-06-27 22:30:19 +00:00
Tim-Philipp Müller
f637e3b80c gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
2007-06-27 12:55:20 +00:00
Tim-Philipp Müller
087f644cfe ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
2007-06-27 10:14:03 +00:00
Wim Taymans
4e676414f0 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Don't error out on 0-sized packets, just emit a warning because this is
not a fatal error. Fixes #451145.
2007-06-27 09:49:51 +00:00
Stefan Kost
757e358d24 docs/plugins/: Update docs with caps info.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins.args:
* 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-decodebin2.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-playbin.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-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 docs with caps info.
2007-06-25 12:43:01 +00:00
Tim-Philipp Müller
17fff87885 po/POTFILES.in: Add more files with translatable strings (#450875).
Original commit message from CVS:
* po/POTFILES.in:
Add more files with translatable strings (#450875).
2007-06-25 12:04:15 +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
Jan Schmidt
dd65a6d76e MAINTAINERS: Updating all the maintainers files
Original commit message from CVS:
* MAINTAINERS:
Updating all the maintainers files
2007-06-22 14:25:27 +00:00
Stefan Kost
62204482e5 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
(main):
Destroy and recreate parse-launch based pipeline after stop to be able
to play again. Reorder some code and add more comments.
2007-06-21 08:34:46 +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
Andy Wingo
ae6fd1b3f2 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
2007-06-19 19:13:04 +00:00
Michael Smith
ba06a86e01 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
2007-06-19 09:34:35 +00:00
Thomas Vander Stichele
f1105b2d06 po/vi.po: Update translations.
Original commit message from CVS:
* po/vi.po:
Update translations.
2007-06-17 17:27:09 +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
Michael Smith
e1cc846edc tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
(eos_buffer_probe):
Add a test that ensures we set DELTA_UNIT on all non-header,
non-video buffers, if we have a video stream.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_process_best_pad):
Move setting delta_pad to earlier, where we inspect all pads, so
that leading audio pages don't get DELTA_UNIT unset if they come
before the first DELTA_UNIT from video pages. Fixes the newly-added
test. Fixes #385527.
2007-06-15 11:15:28 +00:00
Tim-Philipp Müller
67131eaadb tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
fails on the p5-ppc64 build bot and the failure looks like it is due
to the same issue as #348114, ie. a compiler bug.
2007-06-14 19:53:27 +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
34dd1db5f3 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
Fix compilation on mingw. Fixes #446972.
2007-06-13 09:01:32 +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
Sébastien Moutte
a6d8c4109e gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
2007-06-07 21:08:38 +00:00
Tim-Philipp Müller
6f25fde218 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
The BOS page of the first Dirac video stream needs to come before
the BOS page of any Vorbis streams or other audio streams, just like
it is with Theora.
2007-06-07 14:25:32 +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
Wim Taymans
b7c415e7cb configure.ac: check for large file support.
Original commit message from CVS:
* configure.ac:
check for large file support.
2007-06-06 08:01:42 +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
Edward Hervey
b4a04a7057 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
Original commit message from CVS:
Patch by: Edward Hervey  <edward@fluendo.com>
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
(gst_v4lsrc_query):
Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
2007-06-05 16:37:09 +00:00
Tim-Philipp Müller
257a20e77a gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
2007-06-05 16:20:44 +00:00
Wim Taymans
9dac555993 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Handle timestamp wraparound.
2007-06-05 16:19:30 +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
c6ecd5bec8 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
consideratly speedup ogg chain detection by not trying to find a base
timestamp for skeleton streams.
2007-06-05 16:02:57 +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
80c1e3d27c Add draft design for forcing keyframes in encoders and implement in theoraenc.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
* ext/theora/theoraenc.c: (theora_enc_sink_event),
(theora_enc_chain):
Add draft design for forcing keyframes in encoders and implement in
theoraenc.
2007-06-05 15:59:00 +00:00
Jan Schmidt
ba42031248 configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
2007-06-05 13:22:18 +00:00
Jan Schmidt
0d64291611 Release 0.10.13 "What's going on?"
Original commit message from CVS:
Release 0.10.13 "What's going on?"
2007-06-05 12:50:24 +00:00
Wim Taymans
d51693e960 gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge  <slomo@circular-chaos.org>
2007-05-31 17:08:58 +00:00
Jan Schmidt
d6ef01a879 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Define inline when needed on win32 builds. Fixes: #441295
2007-05-31 16:36:22 +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
588bc09c33 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_free), (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
(gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h:
Revert commits towards #152864 made so far. We'll pick it up again
after the 0.10.13 release.
2007-05-25 10:07:26 +00:00
Wim Taymans
b2fdf703c9 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
2007-05-24 16:22:23 +00:00
Tim-Philipp Müller
57375cf664 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
2007-05-24 15:16:59 +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
Jan Schmidt
e1cacbdc9e tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
file. Avoids flukes where the input gets typefound to some valid but
useless type.
2007-05-23 15:54:28 +00:00
Tim-Philipp Müller
b0c7ebb4fc tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
Add unit test for gnomevfssink seeking and position reporting for
file:// URIs.
2007-05-22 15:45:19 +00:00
Mark Nauwelaerts
b274e57bff ext/gnomevfs/gstgnomevfssink.*: see #412648.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
* ext/gnomevfs/gstgnomevfssink.h:
Fix position reporting, especially after a seek (from upstream),
see #412648.
2007-05-22 15:30:26 +00:00
Tim-Philipp Müller
1273d02f4b ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
Repair umlaut.
2007-05-22 15:04:41 +00:00
Jan Schmidt
bec7949e8e gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
2007-05-22 11:40:31 +00:00
Mike Smith
cfc4403058 sys/: Fix a locking-order bug I introduced with my changes the other day.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Fix a locking-order bug I introduced with my changes the other day.
Patch by Mike Smith.
2007-05-21 15:32:42 +00:00
Michael Smith
b48b9fdc19 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_data_packet):
Don't look inside 0-length packets (which indicate duplicated
frames)
2007-05-21 15:24:21 +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
Wim Taymans
7ace85992a gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes #415001.
2007-05-21 09:45:28 +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
Marc-Andre Lureau
16b8bd4c49 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
set_option, get_option, _gst_reserved):
Revert reordering functions (keep ABI).
2007-05-18 15:10:08 +00:00
Jan Schmidt
cbc95dfb3d sys/: When we create our own window, indicate that we handle the
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_show_frame):
When we create our own window, indicate that we handle the
WM_DELETE client message from the window manager, so that it won't
kill our window (and our app) along with it. Handle ClientMessage,
post an error on the bus, and close the window. Further buffers
arriving will result in a FlowError because the window has been
destroyed.
Fixes: #393975
Clean up the X event handling loop and make them the same for
both xvimagesink and ximagesink while I'm at it.
2007-05-17 17:35:46 +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
Tim-Philipp Müller
23396338ad configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
Original commit message from CVS:
* configure.ac:
Bump libtheora requirement to 1.0alpha5 for the pixformat check
(also has a .pc file, so we don't need the fallback check any
longer). Fixes #438840.
2007-05-17 14:17:17 +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
bd9d834bd8 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
Original commit message from CVS:
* docs/Makefile.am: Install docs even when --disable-gtk-doc
is disabled.  This matches the behavior of gtk+.  Fixes #349099.
2007-05-15 20:14:06 +00:00
Wim Taymans
f8f9935d74 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Some more chained streaming ogg timestamp fixes.
2007-05-15 17:11:09 +00:00
Wim Taymans
8b90454ed7 ext/ogg/gstoggdemux.c: Add some FIXMEs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page):
Add some FIXMEs.
Fix chain start/stop segment handling based on patch by
<ahalda at cs dot mcgill dot ca> see #320984.
2007-05-15 16:46:10 +00:00
Michael Smith
171fb33da4 configure.ac: We don't require a C++ compiler. So don't require one.
Original commit message from CVS:
* configure.ac:
We don't require a C++ compiler. So don't require one.
2007-05-15 15:33:54 +00:00
Stefan Kost
38da64193b ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
2007-05-15 15:29:17 +00:00
Marc-Andre Lureau
f2df2a6948 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
gst_alsa_mixer_element_interface_supported,
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
gst_alsa_mixer_element_set_property,
gst_alsa_mixer_element_get_property,
gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
volume_changed, option_changed, _gst_reserved):
Implement notification for alsamixer. Fixes #152864
2007-05-15 14:01:26 +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
David Schleef
1db63972f0 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some sanity checking for the XVImage size returned by X.
Related to #377400.
2007-05-13 01:06:19 +00:00
Wim Taymans
01b6f0b353 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
2007-05-12 16:18:39 +00:00
Wim Taymans
8532e91e7e ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_collect_chain_info):
If there is a stream in a chain without any data packets, ignore the
stream in the total length calculations. Might be related to #436820.
2007-05-12 16:16:22 +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
Michael Smith
9b7339170b ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
gst_pad_push_event() does not return a GstFlowReturn!
2007-05-09 11:54:32 +00:00
Wim Taymans
a8e8bb44d6 tests/examples/seek/: Some small cosmetic changes.
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (stop_cb), (main):
* tests/examples/seek/seek.c: (do_seek):
Some small cosmetic changes.
2007-05-09 11:25:34 +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
Michael Smith
db624febb8 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_type_packet), (theora_dec_change_state):
Don't push events (newsegment, tags) before initialising the
decoder.
This is neccesary for seeking to work correctly in gnonlin.
2007-05-07 11:43:31 +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
Tim-Philipp Müller
9de5f965ea tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Minimal check for volume's GstController usability; also another
test for #422295.
2007-05-04 12:41:21 +00:00
Tim-Philipp Müller
4f0e7a9ef9 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix it so that it (a) makes sense and (b) doesn't break
everything cdda-related including the unit test.
2007-05-04 09:06:38 +00:00
Stefan Kost
57301524fb gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix build when disabling asserts.
2007-05-04 08:46:59 +00:00
Tim-Philipp Müller
cb73a6e792 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
When XShm is not available, we might get row strides that are not
rounded up to multiples of four; this is bad, because virtually
every RGB-processing element in GStreamer assumes rowstrides are
rounded up to multiples of four, so let's allocate at least enough
memory to avoid crashes in this case. The image will still be
displayed distorted though if this happens, so that still needs
fixing (maybe by allocating a bigger image with an 'even' width
and then clipping it appropriately when rendering - something for
Xlib aficionados in any case).
2007-05-03 16:29:10 +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
14f2bca596 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
Original commit message from CVS:
* tests/check/elements/videorate.c: (GST_START_TEST):
Set buffer timestamp to a valid value in order to test the buffer
really does stay in videorate.
2007-05-03 11:24:00 +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
Wim Taymans
7033e458ca gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
Fix offset bug in generation RR packets.
2007-04-29 14:38:05 +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
Wim Taymans
f23356bd8f gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
2007-04-27 15:01:40 +00:00
Tim-Philipp Müller
9e873a3c83 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
2007-04-25 08:54:34 +00:00
Wim Taymans
f5c743b069 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
(read_packet_header), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
(gst_rtcp_packet_sdes_get_item_count),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_entry),
(gst_rtcp_packet_sdes_next_entry),
(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement code to write SR, RR and SDES packets.
2007-04-25 08:10:26 +00:00
Christian Kirbach
80b16b3a4e sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
Original commit message from CVS:
Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
* sys/ximage/ximagesink.c:
Fix build if XShm is not available (#432362).
2007-04-24 20:45:24 +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
478fd77793 ChangeLog surgery: add API keyword
Original commit message from CVS:
ChangeLog surgery: add API keyword
2007-04-21 15:29:27 +00:00
Olivier Crete
e3ff444d30 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
2007-04-21 15:25:22 +00:00
Tim-Philipp Müller
71d77fbecc Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
2007-04-21 15:10:25 +00:00
Zeeshan Ali
80ebb9eb42 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
Original commit message from CVS:
Patch by: Zeeshan Ali  <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
2007-04-21 14:40:45 +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
Tim-Philipp Müller
e832f799e1 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
Original commit message from CVS:
* tests/check/elements/audioresample.c:
Add unit test for audioresample shutdown crasher (#420106).
2007-04-21 13:54:39 +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
Michael Smith
38db14cb22 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
(theora_enc_sink_event), (theora_enc_change_state):
Track initialisation state; don't try to use encoder state if we're
not initialised (it'll segfault).
2007-04-19 16:58:53 +00:00
Stefan Kost
674dcc1b4e tests/check/pipelines/.cvsignore: Fix build.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Fix build.
2007-04-18 11:06:42 +00:00
Sebastian Dröge
9502a0e1b7 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Allow random depths between 1 and 32 instead of only multiplies of 8.
2007-04-17 02:53:16 +00:00
Sebastian Dröge
cef23b669c gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set the maximum number of channels for PCM and float in the correct
place to have it also used when creating the template caps.
2007-04-17 02:04:21 +00:00
Sebastian Dröge
3effb4d23e gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Correctly support 4, 6 and 8 channels with normal PCM and float
wav files.
Fix the depth and signedness calculation in extensible wav files and
also handle 1, 2, 4, 6, 8 channels here when a file without channel
mask is found.
Add support for float, alaw and mulaw in extensible wav files.
This allows correct playback of all but 5 files from
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
(gst_riff_create_audio_template_caps):
Add voxware and float formats to the template caps.
2007-04-17 01:56:07 +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
096abcf3d6 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
Original commit message from CVS:
* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
Don't use pad_alloc_buffer_and_set_caps to create a small header
packet, or, worse, to create a big temporary video buffer using the
src pad.
2007-04-15 14:35:53 +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
631e6435a0 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
streamheader_suite):
Add another test set up for failure
2007-04-13 21:55:31 +00:00
Thomas Vander Stichele
57ba080628 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
GST_START_TEST, streamheader_suite, main):
Add a test for the streamheader bug Wim fixed.
2007-04-13 21:08:11 +00:00
Jan Schmidt
9b2d949a0f ext/theora/theoradec.c: Fix misleading comment.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix misleading comment.
2007-04-13 11:42:34 +00:00
Stefan Kost
95ef089dc6 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
More sanity checks for the header fields.
2007-04-13 06:17:45 +00:00
Tim-Philipp Müller
83ab98b0fc gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Try encodings from all environment variables, not just those in the
first environment variable that is set.
2007-04-12 16:36:36 +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
Michael Smith
cda0d2dc94 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Use scale functions to avoid overflow when calculating duration of
vorbis buffers.
2007-04-12 12:57:33 +00:00
Tim-Philipp Müller
a208469078 API: add gst_tag_freeform_string_to_utf8() (#405072).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
API: add gst_tag_freeform_string_to_utf8() (#405072).
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
Use gst_tag_freeform_string_to_utf8() here.
2007-04-12 12:19:20 +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
Wim Taymans
34a49a9a06 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink),
(gst_play_bin_change_state):
Activate sync in playbin, we are ready to handle it for live streams.
2007-04-06 12:58:06 +00:00
Tim-Philipp Müller
0c9fa8366b tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
Original commit message from CVS:
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream), (playbin_suite):
Add small test for stream-info-value-array code paths.
2007-04-06 09:56:18 +00:00
Wim Taymans
b802dea831 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
2007-04-05 15:44:40 +00:00
Tommi Myöhänen
32a727628f gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstplaybasebin.c: (add_stream):
Fix leak in add_stream(), when g_value_set_object() increases the
refcount of streaminfo object. Fixes #426250.
2007-04-05 10:27:06 +00:00
David Schleef
e859791a21 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a test pattern called "circular", which has concentric
rings with varying radial frequency.  The main purpose of this
pattern is to test fidelity loss in a filter or scaler element.
Notably, this pattern is scale invariant, and is optimally viewed
with a width (and height) of 400.
2007-04-04 02:45:03 +00:00
Tommi Myöhänen
8676f3dce7 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
(deactivate_free_recursive):
Decodebin2 doesn't unref pads it obtains in some occasions:
- multiqueue src pads, when either connecting further or exposing
- sink pads of new autoplugged elements
- peer pads when recursively freeing elements
Fixes #425455.
2007-04-03 11:10:52 +00:00
Sebastian Dröge
fac74a841b gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add audio/x-raw-float support, now that audioconvert support
non-native endianness floats.
2007-03-30 17:05:23 +00:00
Tim-Philipp Müller
90aa33ce83 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
gstreamer-plugins-base.pc doesn't exist, it's
gstreamer-plugins-base-0.10.pc.
2007-03-30 15:00:49 +00:00
René Stadler
6ac8ff9ec3 with some minor changes
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
2007-03-29 18:42:34 +00:00
Wim Taymans
76462ceb45 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_base_init),
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add Private structure.
Bring element code to 2007.
Parse clock-base caps param and use it when generating the
newsegment.
Reset variables before going to PAUSED.
Fix some docs.
2007-03-29 16:23:53 +00:00
Wim Taymans
0a39f494b5 Add RTCP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
Sebastian Dröge
dfdd873f6a gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
PCM samples with width=8 must be always unsigned, no matter what
depth they have.
2007-03-29 12:07:02 +00:00
Andy Wingo
af17f81a47 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
Original commit message from CVS:
2007-03-29  Andy Wingo  <wingo@pobox.com>

* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
perfect offsets also, not just timestamps.

* tests/check/elements/videorate.c (test_more): Test that given
any incoming offsets, that videorate produces perfect offsets.
2007-03-29 11:24:47 +00:00
Wim Taymans
d4015266aa gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add some more RIFF formats.
2007-03-29 10:19:45 +00:00
Wim Taymans
804e7d1759 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix fixed payload names and docs.
Added method to get the default clock rates of fixed payload types.
API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
2007-03-29 10:17:52 +00:00
Zaheer Abbas Merali
01038e30ab tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Add new vorbisdec test to cvsignore.
2007-03-28 15:24:40 +00:00
Wim Taymans
450030ebaf gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-28 14:50:47 +00:00
Sebastian Dröge
293a9c09b8 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes #420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
2007-03-27 12:44:14 +00:00
Michael Smith
e1544977a6 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
2007-03-27 11:31:17 +00:00
Tim-Philipp Müller
726f2c1732 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
2007-03-27 10:17:16 +00:00
René Stadler
01a1e4bc81 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
Original commit message from CVS:
Patch by: René Stadler  <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes #423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes #423055).
2007-03-26 22:38:19 +00:00
Thomas Vander Stichele
ecab77b7e4 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
vorbis_handle_data_packet):
Correctly set DURATION to generate a timestamp-continuous stream.
One bug left at the end; see
ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
* tests/check/Makefile.am:
* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
Add a test to check this.  Without the above patch this test fails.
2007-03-26 20:56:35 +00:00
Jan Schmidt
77683331e1 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
2007-03-26 11:44:07 +00:00
Michael Smith
b3827533a7 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_reset), (gst_video_rate_chain):
If videorate changes caps, we can no longer use the old buffer
(which may have a different size, incompatible with our caps).
So don't do that; just duplicate the new frame more times.
2007-03-23 12:32:33 +00:00
Jan Schmidt
9cbead077e gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
Remove playbin's override of the set_clock vmethod. It's irrelevant
after Wim's commit on the 19th.
2007-03-22 17:43:52 +00:00
Wim Taymans
ffea638f12 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Don't cache file sizes. Fixes #341078.
2007-03-22 09:26:02 +00:00
Tim-Philipp Müller
5b1cd74011 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink):
Use GST_PTR_FORMAT to log caps.
2007-03-21 11:03:23 +00:00
Young-Ho Cha
77cf4f207c gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Special-case some more colour names that pango doesn't handle by
default. Fixes #420578.
2007-03-21 10:23:11 +00:00
Michael Smith
45b6d734ec ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
If we get a zero-sized input buffer, don't pass it to libvorbis, as
that marks EOS internally. After that, libvorbis will buffer all
input data, and encode none of it, eventually leading to memory
exhaustion.
2007-03-20 11:49:55 +00:00
Wim Taymans
d24780a03b gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink):
Don't post STATE_DIRTY anymore.
* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Remove stream_time reset in seek handling, core does that now.
Disable clocking for live pipelines by forcing a NULL clock to the
complete pipeline, core is too smart now for our previous hack.
We can always autoplug in PAUSED now.
2007-03-19 10:52:50 +00:00
David Schleef
819e097960 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
Original commit message from CVS:
* REQUIREMENTS:  Update this file, change the formatting to make
it more consistent, plus more machine readable.
2007-03-18 03:14:01 +00:00
Michael Smith
3bc107dd77 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
2007-03-16 17:29:09 +00:00
Michael Smith
5759241eb4 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
2007-03-16 16:42:23 +00:00
Michael Smith
4ab2d699fd gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.
2007-03-15 10:52:21 +00:00
Philippe Kalaf
b6d7f65463 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes #415001

Indentation/whitespace/documentation fixes.
2007-03-14 21:11:18 +00:00
Julien Moutte
6940042ecf gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
Original commit message from CVS:
2007-03-14  Julien MOUTTE  <julien@moutte.net>

* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.
2007-03-14 17:16:30 +00:00
Thomas Vander Stichele
8b80c6f13a po/: Update translations from translation project.
Original commit message from CVS:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update translations from translation project.
2007-03-14 15:16:23 +00:00
Thomas Vander Stichele
081deac039 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample
2007-03-14 14:48:12 +00:00
Michael Smith
7292973429 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
Since the plugin doesn't support anything other than 4:2:0 right
now, post an error and fail if we get something else. Won't matter
until libtheora supports the other pixel formats, but hopefully
that'll be soon...
2007-03-12 23:29:07 +00:00
Alex Lancaster
7931b94d65 I'm too lazy to comment this
Original commit message from CVS:
Mention Patch by: Alex Lancaster in a recent commit.
2007-03-12 15:50:35 +00:00
Sébastien Moutte
1596dd263c gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
2007-03-10 15:59:33 +00:00
Tim-Philipp Müller
4462906be4 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes #410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
2007-03-10 12:18:58 +00:00
Tim-Philipp Müller
5de913c273 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
Original commit message from CVS:
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add unit test for MPL2 subtitle format (#413799).
2007-03-10 11:21:08 +00:00
Kamil Pawlowski
389eb6f2f8 gst/subparse/: Add support for MPL2 subtitle format (#413799).
Original commit message from CVS:
Patch by: Kamil Pawlowski  <kamilpe gmail com>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
* gst/subparse/mpl2parse.h:
Add support for MPL2 subtitle format (#413799).
2007-03-10 11:17:52 +00:00
Tim-Philipp Müller
135568e80b configure.ac: We require core CVS for the new buffer metadata copy functions.
Original commit message from CVS:
* configure.ac:
We require core CVS for the new buffer metadata copy functions.
2007-03-09 17:33:17 +00:00
Wim Taymans
5676bdaf81 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Fixes #414496.
2007-03-09 16:51:13 +00:00
Wim Taymans
5d5f935e38 ext/libvisual/visual.c: Improve adapter usage and comments.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
Improve adapter usage and comments.
2007-03-09 16:46:35 +00:00
Wim Taymans
e9be846621 Use new metadata copy function.
Original commit message from CVS:
* ext/pango/gsttextrender.c: (gst_text_render_chain):
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
Use new metadata copy function.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
Basetransform copied the metadata for us.
2007-03-09 16:38:06 +00:00
Tim-Philipp Müller
b9ca5573a9 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
(gst_text_overlay_video_event):
Some more logging. Only accept newsegment events in TIME format and
send a WARNING message if they are not in TIME format.
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
(gst_sub_parse_chain), (gst_sub_parse_sink_event):
* gst/subparse/gstsubparse.h:
No need to allocate GstSegment structure dynamically, just put it
into the instance structure; ignore newsegment events in BYTE
format and in particular don't let it overwrite our saved TIME
segment from the last seek.
2007-03-09 16:28:04 +00:00
Michael Smith
c9cd89bbab gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
Replace AC3 typefinder with one that isn't terrible, and actually
works usefully.
2007-03-09 13:05:04 +00:00
Thomas Vander Stichele
e81b9ec719 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
fix error category and translatable string
2007-03-09 12:22:53 +00:00
Tim-Philipp Müller
6588812aaf pkgconfig/: Fix up utils => pbutils here too.
Original commit message from CVS:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Fix up utils => pbutils here too.
2007-03-09 11:23:32 +00:00
Tim-Philipp Müller
3eba8b10e2 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer):
Break out of loop in chain function as soon as possible if we get
a non-OK flow return.
2007-03-09 10:49:53 +00:00
Jan Schmidt
9d1fc287e0 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
Original commit message from CVS:
* tests/check/elements/alsa.c: (GST_START_TEST):
Unref the mixer if the state change fails too (if the
alsa devices are inaccessible, for example)
2007-03-08 18:26:07 +00:00
Jan Schmidt
9c26f99cce tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
Original commit message from CVS:
* tests/check/Makefile.am:
Don't test libvisual elements in the states check, because libvisual
seems to leak internally.
Re-enable the alsa and states tests now that there's new suppressions
in gst.supp.
* tests/check/elements/alsa.c: (GST_START_TEST):
Don't leak the alsamixer we instantiated.
2007-03-08 17:49:46 +00:00
Jan Schmidt
ea0d6ebf36 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset),
(gst_ximagesink_finalize):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
Move some cleanup stuff from the state change handler into a _reset()
function that can be called from _finalize(). This ensures that things
get freed even if (for some reason) the NULL->READY state transition
fails in the parent class.
Even if a parent state change fails, process our downward state change
logic instead of bailing out early.
Free the correct xcontext pointer in ximagesink's xcontext_clear.
2007-03-08 15:22:53 +00:00
Jan Schmidt
46a7bd8dfd ext/alsa/gstalsasink.c: Extra log line.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
Extra log line.
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
Use pango_font_description_set_family_static instead of
pango_font_description_set_family to save a string copy (it was
leaking due to the strdup anyway)
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
Chain up in finalize.
2007-03-08 12:53:51 +00:00
Tim-Philipp Müller
4aa8b0ca21 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: add "untranslated-label" property which should be set by
implementations at construct time (#414645).
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set "untranslated-label" when constructing mixer track objects.
* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
Unit test to check the above.
2007-03-07 18:50:10 +00:00
Wim Taymans
a1c380650c ext/ogg/gstoggdemux.c: Fix confusing debug message.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
Fix confusing debug message.
2007-03-07 17:15:57 +00:00
Jan Schmidt
0380355662 gst-plugins-base.doap: update doap file with new version
Original commit message from CVS:
* gst-plugins-base.doap:
update doap file with new version
2007-03-07 17:12:54 +00:00
Jan Schmidt
f8534cc908 configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
2007-03-07 16:56:01 +00:00
Jan Schmidt
c84f76e4b4 Release 0.10.12
Original commit message from CVS:
Release 0.10.12
2007-03-07 16:46:51 +00:00
Jan Schmidt
8a9f908ea9 configure.ac: Bump version to 0.10.11.4 pre-release
Original commit message from CVS:
* configure.ac:
Bump version to 0.10.11.4 pre-release
2007-03-06 12:31:01 +00:00
Wim Taymans
a2a8b1b8ce gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes #414684.
2007-03-06 12:10:08 +00:00
Jan Schmidt
d7e51be66f configure.ac: Bump version to 0.10.11.3 pre-release
Original commit message from CVS:
* configure.ac:
Bump version to 0.10.11.3 pre-release
2007-03-05 11:21:13 +00:00
Sebastian Dröge
80fe97c1f8 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
Original commit message from CVS:
* po/POTFILES.in:
Update paths for the rename from utils to pbutils to fix the build.
2007-03-05 09:35:29 +00:00
Tim-Philipp Müller
5d14dbbcda gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
Change directory to install headers in from gst/utils to gst/pbutils
as well.
2007-03-05 09:27:55 +00:00
Thomas Vander Stichele
f6bd20e5e3 rename utils to pbutils
Original commit message from CVS:
* configure.ac:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
(gst_pb_utils_get_source_description),
(gst_pb_utils_get_sink_description),
(gst_pb_utils_get_decoder_description),
(gst_pb_utils_get_encoder_description),
(gst_pb_utils_get_element_description),
(gst_pb_utils_add_codec_description_to_tag_list),
(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
* gst-libs/gst/pbutils/descriptions.h:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_source_message_new),
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new),
(gst_missing_plugin_message_get_description):
* gst-libs/gst/pbutils/missing-plugins.h:
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/pbutils/pbutils.h:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/descriptions.h:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/install-plugins.h:
* gst-libs/gst/utils/missing-plugins.c:
* gst-libs/gst/utils/missing-plugins.h:
* gst-plugins-base.spec.in:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
* tests/check/Makefile.am:
* tests/check/libs/pbutils.c: (GST_START_TEST),
(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
* tests/check/libs/utils.c:
rename utils to pbutils
2007-03-04 23:39:51 +00:00
Jan Schmidt
c73e88a6a1 Add documentation for decodebin2 that indicates that the API is still unstable.
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:
* docs/plugins/inspect/plugin-decodebin2.xml:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
Add documentation for decodebin2 that indicates that the API
is still unstable.
2007-03-02 12:59:15 +00:00
Jan Schmidt
d11000959b configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
Original commit message from CVS:
* configure.ac:
Update to 0.10.11.2 (0.10.12 pre-release)
2007-03-01 18:50:00 +00:00
Wim Taymans
5ee0a694a6 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
base time is irrelevant here.
2007-03-01 17:29:55 +00:00
Wim Taymans
85c7eeecc3 gst-libs/gst/audio/: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Improve debugging.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Improve latency and clock slaving calculations.
Improve slave clock calibration.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
When we are asked to render N sample to 0 bytes, return N.
2007-03-01 17:01:43 +00:00
Wim Taymans
1d0e1586d8 ext/alsa/gstalsasink.*: Remove unused dispose function.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
(gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Remove unused dispose function.
Rename lock to not interfere with alsasrc lock.
* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
(gst_alsasrc_read), (gst_alsasrc_reset):
* ext/alsa/gstalsasrc.h:
Implement finalize function.
Use lock to protect alsa access.
Implement _reset.
Fine tune sw params.
2007-03-01 16:48:45 +00:00
Thomas Vander Stichele
acdd48757f configure.ac: Convert to new AG_GST style.
Original commit message from CVS:
* configure.ac:
Convert to new AG_GST style.
2007-02-28 19:27:28 +00:00
Ed Catmur
61eaa5eb9b gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
Original commit message from CVS:
Patch by: Ed Catmur <ed at catmur dot co dot uk>
* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
Fix race condition when rapidly switching visualisations in playbin.
Fixes #401029.
2007-02-28 15:17:20 +00:00
Jan Schmidt
91e7d53d42 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
Original commit message from CVS:
* tests/check/Makefile.am:
Include local stuff before system installed things in LDFLAGS and
CFLAGS.
2007-02-28 15:11:59 +00:00
Wim Taymans
ce47c4d2d8 ext/ogg/gstoggdemux.c: Improve debugging.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
Improve debugging.
2007-02-28 15:10:06 +00:00
Wim Taymans
f1f26384e9 sys/v4l/: Fix duration and timestamping, taking latency into account.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
(gst_v4lsrc_fixate), (gst_v4lsrc_query):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
Fix duration and timestamping, taking latency into account.
Implement latency query.
2007-02-28 15:05:03 +00:00
Wim Taymans
3c94c06c5a gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.
2007-02-28 15:02:25 +00:00
Thomas Vander Stichele
eb6d2476c6 tests/check/generic/states.c: Copy the states.c test from core again
Original commit message from CVS:
* tests/check/generic/states.c: (GST_START_TEST):
Copy the states.c test from core again
* tests/check/Makefile.am:
ignore cdio and cdparanoiasrc
2007-02-28 12:44:53 +00:00
Stefan Kost
14cd6e6ae1 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index), (check_default),
(audio_convert_prepare_context), (audio_convert_convert):
Also make valgrind happy and avoid copying data in some cases.
2007-02-28 12:08:27 +00:00
Stefan Kost
2c67c89457 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes #339837 even more.
2007-02-28 11:47:45 +00:00
Julien Moutte
354c07de8b tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
Original commit message from CVS:
2007-02-27  Julien MOUTTE  <julien@moutte.net>

* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
(msg_segment_done): Fix various seeking bugs (Slider was not
updating when doing a non flushing seek, Reverse playback
on segment seek was wrong).
2007-02-27 18:45:37 +00:00
Wim Taymans
6c8b08b09a tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_seek):
When we stop scrubbing, don't leave the pipeline PLAYING when we
requested a PAUSED state.
2007-02-26 11:48:49 +00:00
René Stadler
88e94fc278 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
2007-02-25 23:51:03 +00:00
Loïc Minier
2770d0fbea Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
2007-02-24 20:12:49 +00:00
Tim-Philipp Müller
d35f4efe67 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Improve docs: point out that the application needs to assist playbin
with buffering.
2007-02-23 18:49:29 +00:00
Tim-Philipp Müller
e8e648a76d Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
2007-02-23 13:10:50 +00:00
Tim-Philipp Müller
011471dbbb gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
2007-02-22 12:57:47 +00:00
Andy Wingo
d9b6796d91 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
2007-02-22 11:04:10 +00:00
Stefan Kost
00d7c52de8 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes #339837
2007-02-22 09:04:37 +00:00
Jan Schmidt
841750c80a tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek):
Undo the previous commit: -1 as a stop time implies that the stop
time is the end of file, clearing any previously configured segment.
2007-02-21 16:12:15 +00:00
Jan Schmidt
2d71ddc3c9 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek):
Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
2007-02-21 15:36:26 +00:00
Stefan Kost
1b01b41b69 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps):
Unbreak volume, value remains gint.
2007-02-21 13:55:54 +00:00
Stefan Kost
b8dfedde6a gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_double), (volume_process_float),
(volume_process_int16), (volume_process_int16_clamp),
(volume_set_caps), (volume_transform_ip), (volume_update_volume):
* gst/volume/gstvolume.h:
Extend float audio support (double) and some int->uint cleanups.
2007-02-21 13:08:51 +00:00
Edward Hervey
7bc6fc07a1 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
(sort_end_pads), (gst_decode_group_expose),
(gst_decode_group_hide):
Don't free groups from the streaming threads. Just put them aside and
free them in dispose.
2007-02-20 15:44:32 +00:00
Edward Hervey
7746c2f176 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_element),
(pad_added_group_cb), (gst_decode_group_check_if_blocked),
(sort_end_pads), (gst_decode_group_expose):
Handle dynamic pads within groups.
Sort pads before exposing them in order to make playbin happy.
There still is a race with the multiqueue filling up. This should be
solved separately.
Fixes #398721
2007-02-20 11:20:52 +00:00
Tim-Philipp Müller
b99629643c gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
Original commit message from CVS:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
Some more docs (and descriptions for two subtitle formats).
2007-02-18 21:02:36 +00:00
Tim-Philipp Müller
2f45e10c73 gst-libs/gst/audio/audio.c: Fix documentation.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix documentation.
2007-02-16 10:19:45 +00:00
Yves Lefebvre
a6e3e71fe3 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
Original commit message from CVS:
Patch by: Yves Lefebvre  <ivanohe abacom com>
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
Don't leak caps. Fixes #408278.
2007-02-16 10:15:46 +00:00
Stefan Kost
b2f9c0f289 More docs coverage and some ChangeLog surgery (add missing names)
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.h:
* ext/ogg/gstoggdemux.h:
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst/adder/gstadder.h:
More docs coverage and some ChangeLog surgery (add missing names)
2007-02-15 15:17:23 +00:00
Wim Taymans
6ed39cf7c7 sys/: Small constifications.
Original commit message from CVS:
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio):
Small constifications.
2007-02-15 12:07:57 +00:00
Wim Taymans
a43d0f57eb gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 12:06:25 +00:00
Wim Taymans
70e52caf04 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
Ignore errors in reset, these are not fatal. They also grab the element
lock which is already taking when this function is called. Fixes
#405451.
2007-02-15 11:59:41 +00:00
Stefan Kost
397eb100e9 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
Original commit message from CVS:
* configure.ac:
Remove 'tests/examples/xerror/Makefile' from output files again.
2007-02-13 10:24:13 +00:00
Stefan Kost
cfc36dbcb8 Also crossref against gst-plugins-base-libs.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Also crossref against gst-plugins-base-libs.
2007-02-13 09:12:11 +00:00
Stefan Kost
7ee1b714f0 Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
* gst-libs/gst/audio/audio.h:
Source formatting.
* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
Add own debug category.
2007-02-12 20:42:23 +00:00
René Stadler
14f2d7efdb gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
(#403597).
2007-02-12 11:01:04 +00:00
Tim-Philipp Müller
a58a6b1854 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
When we have external subtitles and wait for the subtitle decodebin
to get up and running, we set up a (sync) bus handler for the
subtitle decodebin, so we can stop waiting when it posts an error
message. However, we should do that before we set the subtitle
decodebin's state to playing, otherwise things are racy and we might
miss error messages posted before we had a chance to set up the bus.
This should finally fix totem hanging on .txt pseudo-subtitle files.
2007-02-12 10:33:40 +00:00
Sébastien Moutte
9caee48ed4 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Use gst_gdouble_to_guint64 for conversions.
* win32/common/config.h.in:
Add a define for GST_INSTALL_PLUGINS_HELPER
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstrtp.def:
* win32/common/libgutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstnetbuffer.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstvorbis.dsp:
* win32/vs6/libgstcdda.dsp:
* win32/vs6/libgstgdp.dsp:
* win32/vs6/libgstutils.dsp:
Update and add new project files.
2007-02-10 19:27:48 +00:00
Tim-Philipp Müller
5f186351a1 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
(subrip_remove_unhandled_tags), (parse_subrip):
For SubRip (.srt) subtitles, ignore all markup tags we don't
handle (like font tags, for example).
* tests/check/elements/subparse.c:
Add test for this.
2007-02-10 18:19:37 +00:00
Tim-Philipp Müller
d2f03d2cec ChangeLog surgery
Original commit message from CVS:
ChangeLog surgery
2007-02-09 13:28:01 +00:00
Tim-Philipp Müller
12f5dd47ef gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (add_fakesink),
(gst_decode_bin_change_state):
Don't error out if there is no fakesink in the READY to NULL state
change, since when decodebin is re-used, we're only adding the
fakesink element in READY to PAUSED.
* tests/check/elements/decodebin.c:
(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
(decodebin_suite):
Minimal unit test to make sure we can use the same decodebin
instance twice (at least with audiotestsrc input).
2007-02-09 13:16:27 +00:00
Tim-Philipp Müller
662c557cc2 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
Try to get devic-name from device string first, and from handle only
as fallback (seems to yield better results and is more robust
against buggy probing code on the application side).
2007-02-09 09:58:28 +00:00
Julien Puydt
880da4d8f1 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
Original commit message from CVS:
Based on patch by: Julien Puydt <julien.puydt at laposte net>
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
(gst_alsa_find_device_name):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
Improve device-name detection a bit, especially in the case where
the device is not actually open (#405020, #405024). Move common code
into gstalsa.c instead of duplicating it.
2007-02-08 15:43:26 +00:00
Tim-Philipp Müller
b827e4d539 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
2007-02-07 13:05:01 +00:00
Julien Moutte
bbae92bbc4 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
Original commit message from CVS:
2007-02-06  Julien MOUTTE  <julien@moutte.net>

* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
(gst_xvimagesink_get_xv_support),
(gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_interface_supported),
(gst_xvimagesink_probe_get_properties),
(gst_xvimagesink_probe_probe_property),
(gst_xvimagesink_probe_needs_probe),
(gst_xvimagesink_probe_get_values),
(gst_xvimagesink_property_probe_interface_init),
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
(gst_xvimagesink_init), (gst_xvimagesink_class_init),
(gst_xvimagesink_get_type):
* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
for XVAdaptors so that one can choose the adaptor to use with
gstreamer-properties.
2007-02-06 17:47:32 +00:00
Stefan Kost
66727fe484 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Also mention that a conversion from double to float is suboptimal still.
2007-02-06 14:00:31 +00:00
Tim-Philipp Müller
5b499dec66 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
2007-02-06 09:42:05 +00:00
Andy Wingo
451ff2f992 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
2007-02-05 18:39:51 +00:00
Tim-Philipp Müller
2933302ce8 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
A width and height of 1 makes us crash, so increase minimum size to
2x2 pixels until someone feels like fixing this (#404512).
2007-02-05 11:44:52 +00:00
Tim-Philipp Müller
e7a698506c tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
Original commit message from CVS:
* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
Add small test to make sure request pads are cleaned up properly
even if oggmux never changes state out of NULL.
2007-02-04 16:23:37 +00:00
Tim-Philipp Müller
b1751ae3cb tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
Original commit message from CVS:
* tests/check/libs/utils.c: (GST_START_TEST):
Fix unit test. Turns out things work much better when you
NULL-terminate string arrays. Should make p5 build bot happy again.
2007-02-04 14:11:51 +00:00
Tim-Philipp Müller
2594880e87 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init),
(gst_audio_filter_template_class_init),
(gst_audio_filter_template_init),
(gst_audio_filter_template_set_property),
(gst_audio_filter_template_get_property),
(gst_audio_filter_template_setup),
(gst_audio_filter_template_filter),
(gst_audio_filter_template_filter_inplace), (plugin_init):
Oops, forgot to commit fixed-up example.
2007-02-03 23:28:45 +00:00
Tim-Philipp Müller
b63fff63d4 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
2007-02-03 20:19:35 +00:00
Tim-Philipp Müller
7d78598f24 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_spawn_child):
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Lowering log level to see why things fail on the p5 build bot;
fix some typos in unit test messages.
2007-02-03 14:26:54 +00:00
Tim-Philipp Müller
92c11d5d24 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
Original commit message from CVS:
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Don't hard-code temp directory for test helper; use GLib functions
to write out file and do error checking etc.
2007-02-03 13:59:27 +00:00
Tim-Philipp Müller
17a02da2fd gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
Original commit message from CVS:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_context_set_xid),
(gst_install_plugins_context_new),
(gst_install_plugins_context_free),
(gst_install_plugins_get_helper),
(gst_install_plugins_spawn_child),
(gst_install_plugins_return_from_status),
(gst_install_plugins_installer_exited),
(gst_install_plugins_async), (gst_install_plugins_sync),
(gst_install_plugins_return_get_name),
(gst_install_plugins_installation_in_progress):
* gst-libs/gst/utils/install-plugins.h:
API: add API for applications to initiate installation of missing
plugins, ie. gst_install_plugins_async() primarily.
Based on libgimme-codec by Ryan Lortie.
* configure.ac:
Add --with-install-plugins-helper configure option so distros can specify
the path of the helper script or program to call when plugin installation
is requested (distros: please do any argument munging in this helper
script instead of patching GStreamer to pass arguments differently
to another program directly).
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Build and document new API.
* tests/check/libs/utils.c: (result_cb),
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
(libgstbaseutils_suite):
Some simple checks for the new API.
2007-02-02 20:42:08 +00:00
Tim-Philipp Müller
7439949980 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (test_float_conversion):
Add small test for 32bit float <=> 64bit float conversion (works
only one way so far, 32=>64 produces structured noise).
2007-02-02 14:44:29 +00:00
Tim-Philipp Müller
b7cf10eb6d gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
2007-02-02 11:21:48 +00:00
Stefan Kost
36180a5f3d gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
2007-02-02 09:48:53 +00:00
Holger Wansing
6f25185e75 po/: Add German translation (#352069).
Original commit message from CVS:
Patch by: Holger Wansing  <linux wansing-online de>
* po/LINGUAS:
* po/de.po:
Add German translation (#352069).
2007-02-01 18:50:08 +00:00
Sebastian Dröge
d1778e3f6d ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
Original commit message from CVS:
reviewed by: Wim Taymans <wim@fluendo.com>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
Use newly added GstCollectPads API to free the allocated resources in
the GstOggPad structures (#402393).
2007-02-01 17:52:39 +00:00
Jan Schmidt
800cd83487 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_vis_element):
Add audioresample+audioconvert in front of the visualisation
element, so that elements like libvisual 0.4 that don't support all
samplerates can work.
Fixes: #402505
2007-01-31 15:58:53 +00:00
Tim-Philipp Müller
514c2d7610 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
Take some locks and make a copy of the streaminfo value array we
maintain while holding the lock, so that the application can
retrieve the stream-info as a value array in a thread-safe way.
2007-01-30 19:19:37 +00:00
Wim Taymans
75d5fcb62e gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Don't fail on 0 sized buffers. Fixes #396835.
2007-01-30 11:29:17 +00:00
David Schleef
888ea4730e gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Detect BBCD as video/x-dirac, so we can play raw dirac
streams.
2007-01-29 21:13:07 +00:00
Tim-Philipp Müller
a37616d4df ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
Original commit message from CVS:
* ext/theora/theoraenc.c: (theora_enc_chain):
Check return value of theora_encode_header(), or we might try to
allocate a random number of bytes. theora_encode_header() can fail
if libtheora has been compiled with encoding support disabled.
Fixes #398110.
2007-01-29 18:14:25 +00:00
Wim Taymans
d49920f3a2 tests/check/gst/.cvsignore: Do as buildbot says.
Original commit message from CVS:
* tests/check/gst/.cvsignore:
Do as buildbot says.
2007-01-29 10:53:06 +00:00
Wim Taymans
5be1e44107 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_src_setcaps):
Fix strides in libvisual. Gst uses X strides.
Inspired by: <ed at catmur dot co dot uk> and
<tim at centricular dot net>
Fixes #401118.
2007-01-29 10:25:11 +00:00
Wim Taymans
fde9b0096c ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
(gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
(gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
(gst_ogg_demux_perform_seek),
(gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
(gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
(gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
(gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
* ext/ogg/gstoggdemux.h:
Properly propagate streaming errors when we are scanning the file for
chains so that we don't crash when shut down. Might fix some crashers
when quickly switching oggs in RB such as #332503 and #378436.
2007-01-27 13:32:24 +00:00
Tim-Philipp Müller
3b3320ac38 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
error code as well.
2007-01-26 12:44:46 +00:00
Wim Taymans
ce1ff3c94c gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_source):
Don't try to disconnect a signal from a finalized object.
2007-01-25 16:02:41 +00:00
Tim-Philipp Müller
af52900954 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
Cast lock macro parameters to make sure we're actually accessing the
lock member at the right class level. Free list itself in _dispose()
as well and NULL it in case dispose gets called multiple times.
2007-01-25 14:29:21 +00:00
Edward Hervey
a8d29e1f82 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_bin_dispose),(gst_decode_bin_finalize):
Free GstDecodeGroups no longer used.
(gst_decode_group_expose):
Don't unlock too many times !
(deactivate_free_recursive):
Free iterator once we're done with it.
Fix for recursively deactivating elements (stop at ghostpads).
2007-01-25 14:02:37 +00:00
Tim-Philipp Müller
dab1309f32 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (handoff):
Fix up caps on the frame buffer before we save it and potentially
make it accessible to other threads via g_object_get; also use
gst_buffer_replace() instead of gst_mini_object_replace().
2007-01-25 12:24:18 +00:00
Tim-Philipp Müller
d289e2d55c gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Make getting the current frame thread-safe.
2007-01-25 12:06:59 +00:00
Edward Hervey
77cf01b76b gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
(gst_decode_group_new), (gst_decode_group_free):
Set queues to bigger sizes to cope with HD contents.
Fix some mutex freeing and add comment about MT safe methods.
2007-01-25 11:48:10 +00:00
Tim-Philipp Müller
5ed71164a3 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
(gst_text_overlay_text_event):
Don't unnecessarily ref (and then leak) upstream events if the text
pad is not linked. Fixes #399948.
* tests/check/gst-plugins-base.supp:
Add suppression for pango on edgy/x86 for textoverlay test.
2007-01-24 12:51:20 +00:00
Wim Taymans
81e92118da gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add some more fixed payloads.
2007-01-24 12:10:56 +00:00
Tim-Philipp Müller
c135f896e5 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
Error out properly if we get an error from libogg while reading the
BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
2007-01-23 18:39:45 +00:00
Tim-Philipp Müller
b8d04c8843 gst/playback/gstdecodebin2.c: Don't leak mutex.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
Don't leak mutex.
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_urisource_handler),
(test_missing_suburisource_handler),
(test_missing_primary_decoder), (playbin_suite):
Run all tests once with decodebin and once with decodebin2.
One test does not pass yet with decodebin2.
2007-01-23 17:49:29 +00:00
Edward Hervey
0fcb92f2dd ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
Fix the cases where oggmux doesn't properly figure out that all
sinkpads have gone EOS, and therefore doesn't push out the remaining
buffers and the final EOS event.
Fixes #363379
2007-01-23 14:30:28 +00:00
Julien Moutte
60dad35de0 sys/: Don't lock on navigation event push, just on keysym to string.
Original commit message from CVS:
2007-01-23  Julien MOUTTE  <julien@moutte.net>

* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Don't lock on navigation event push, just on keysym to string.
Fixes #397673 again.
2007-01-23 13:19:19 +00:00
Edward Hervey
a5146c9037 gst/playback/gstdecodebin2.c: Cleanups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(get_current_group), (group_demuxer_event_probe),
(gst_decode_group_expose), (deactivate_free_recursive),
(gst_decode_group_free):
Cleanups.
Don't forget to emit 'no-more-pads' once a group is exposed.
Cleanup elements from a DecodeGroup once we remove it.
Protect call to gst_decode_group_expose() with the decodebin lock.
2007-01-22 17:37:38 +00:00
Julien Moutte
7861683872 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
Original commit message from CVS:
2007-01-22  Julien MOUTTE  <julien@moutte.net>

* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Looking at Xorg code i can't figure out if that XKeysymToString
function is thread sensible or not. Lock it just in case as
recommended by Radek Doulik <rodo at ximian dot com>.
2007-01-22 13:16:42 +00:00
Julien Moutte
34af6e729d sys/: Lock that X Call as well. Fixes #397673.
Original commit message from CVS:
2007-01-22  Julien MOUTTE  <julien@moutte.net>

* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Lock that X Call as well. Fixes #397673.
2007-01-22 13:10:13 +00:00
Tim-Philipp Müller
0da3960238 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
Don't go into an endless loop if the file starts with 00 00 01 2X,
like quicktime redirect files might. Fixes #396042.
* tests/check/Makefile.am:
* tests/check/gst/.cvsignore:
* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
(typefindfunctions_suite):
Add unit test for the above.
2007-01-22 12:03:27 +00:00
Tim-Philipp Müller
58e6e134cb gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
On second thought, use "depth" field rather than "bpp" field.
2007-01-22 10:27:26 +00:00
Tim-Philipp Müller
439b3193bd gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Camtasia caps apparently need a bpp field (#398875).
2007-01-22 09:23:01 +00:00
Tim-Philipp Müller
acf3bcdfbb gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element), (gst_play_base_bin_change_state):
Attempt at a better error message in case we don't have the required
URI handler installed; post missing-plugin message also when we're
missing an URI handler for the subtitle URI; clean up properly also
when an error occurs and we never made it to PAUSED state.
* tests/check/elements/playbin.c: (GST_START_TEST),
(playbin_suite):
Check that we're also getting a missing-plugin messsage for a
missing subtitle URI handler (and clean up properly).
2007-01-19 19:09:05 +00:00
Tim-Philipp Müller
638dbd7b71 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
Plug a few reference leaks.
2007-01-19 18:47:30 +00:00
Tim-Philipp Müller
91773593cf gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Lower probability a bit if the marker isn't right at the start,
to decrease the chance of false positives.
2007-01-19 12:23:06 +00:00
Tim-Philipp Müller
c47074371a gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Small mpeg2 system stream typefinding improvement: make typefinder
probe a bit into the stream instead of just looking for a marker
at the beginning. Fixes #397810.
2007-01-19 11:31:50 +00:00
Tim-Philipp Müller
4936d6ba02 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
Remove compatibility cruft for prehistoric GLib versions.
2007-01-18 16:23:35 +00:00
Tim-Philipp Müller
5262208d6e gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c: (close_pad_link):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_handle_message_func), (unknown_type):
Let decodebin be the element to post missing-plugin messages for
missing decoders (rather than playbin); make playbin implement
GstBin::handle_message so we can suppress missing-plugin messages
for types we're not handling on purpose (don't want to bring up an
installer in those cases).
2007-01-17 16:11:14 +00:00
Tim-Philipp Müller
0eac623115 gst/: Fix potentially unaligned access (#397207).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
Fix potentially unaligned access (#397207).
2007-01-16 19:37:55 +00:00
Stefan Kost
268dcb0ab1 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
Original commit message from CVS:
* tests/examples/seek/seek.c: (set_scale), (update_scale),
(do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
(rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
(main):
Allow to toggle looping while it plays. Fix callback prototype. Clean
up code a bit more. Add copyright header.
2007-01-16 12:17:06 +00:00
Stefan Kost
bc62bb15e5 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
Original commit message from CVS:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Red and blue mask was swapped (spotted by Dan Williams).
2007-01-16 11:41:58 +00:00
Stefan Kost
9f6e8af294 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Use new beats-per-minute tag from core.
2007-01-15 13:58:58 +00:00
Tim-Philipp Müller
dbb27a7c74 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
Original commit message from CVS:
* po/POTFILES.in:
Add new files with translatable strings, so they actually make it
into the template file one day.
2007-01-15 11:30:53 +00:00