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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-31  Stefan Kost  <ensonic@users.sf.net>

	* 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.

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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-30  Stefan Kost  <ensonic@users.sf.net>

	* 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.

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2007-10-30  Stefan Kost  <ensonic@users.sf.net>

	* tests/icles/stress-playbin.c:
	  Fix the build.

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2007-10-30  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-30  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-30  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-28  Sebastian Dröge  <slomo@circular-chaos.org>

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	* gst-libs/gst/riff/riff-media.c:
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	(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.

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2007-10-26  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-25  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-25  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

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2007-10-24  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-10-23  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

	* 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. 

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2007-10-16  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-16  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/stress-playbin.c:
	  Does what it says on the tin.

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2007-10-15  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
	Fix queue negotiation. See #486758.

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2007-10-12  Jan Schmidt  <Jan.Schmidt@sun.com>
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	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
	(gst_xvimagesink_xwindow_new),
	(gst_xvimagesink_update_colorbalance),
	(gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):

	Fix handling of some of the X atoms. If the last parameter is True,
	XInternAtom won't create the atom if it doesn't exist, and therefore
	might return None. This causes X errors on Xv implementations that
	don't provide the colour balance attributes.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c:
	* tests/check/libs/tag.c:
	  Extract vorbis comment LICENSE tags correctly.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	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).

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2007-10-10  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_drain):
	Use new basesink method to make our EOS drain interruptable.

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2007-10-10  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst-libs/gst/rtp/gstrtppayloads.c:
	Fix silly search-replace oversight.

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2007-10-09  Wim Taymans  <wim.taymans@gmail.com>

	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.


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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
	Fix debug output.

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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-08  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst-libs/gst/rtp/gstrtppayloads.c:
	Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
	to avoid compiler warnings

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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gsttagdemux.c:
	  Don't leak caps.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-10-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Replace g_return_if_val (as it could be disabled), with regular return
	  and warning.

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2007-10-03  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/simple-launch-lines.c:
	  Print message name and not just number.

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2007-10-02  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/snapshot/snapshot.c:
	  Fix the build.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/sdp/gstsdpmessage.h:
	Add RFC 3556 bandwidth modifiers.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtppayloads.c:
	Update documentation.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-29  Sebastien Moutte <sebastien@moutte.net>

	* 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.
	
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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/snapshot/snapshot.c: (main):
	Print error when pipeline failed to construct.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.
	  
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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
	of duplicating these macros in configure.ac.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/sv.po:
	* po/uk.po:
	  Updated translations to 0.10.14

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jakub Bogusz <qboosh@pld-linux.org>

	* po/pl.po:
	  Added Polish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Ilkka Tuohela <hile@iki.fi>

	* po/fi.po:
	  Added Finnish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jorge González González <aloriel@gmail.com>

	* po/es.po:
	  Added Spanish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mogens Jaeger <mogens@jaeger.tf>

	* po/da.po:
	  Added Danish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Funda Wang <fundawang@linux.net.cn>

	* po/zh_CN.po:
	  Added Chinese (simplified) translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Alexander Shopov <ash@contact.bg>

	* po/bg.po:
	  Added Bulgarian translation.

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2007-09-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstqueue2.c: (gst_queue_push_one):
	Fix compilation wrt printf arguments.

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2007-09-20  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-20  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/generic/states.c:
	  Improved state change unit test.

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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/.cvsignore:
	* tests/check/.cvsignore:
	  Ignore registries in any format.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* ext/gnomevfs/gstgnomevfssrc.c:
	  Improve debug logs a bit and be more verbose if things go wrong.

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2007-09-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* 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.

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2007-09-17  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-15  David Schleef  <ds@schleef.org>

	* 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.

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2007-09-15  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
	Add some debug info when negotiating caps.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	A buffer with an empty payload is also a valid buffer.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-14  Julien MOUTTE  <julien@moutte.net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
	typefind for QCP files (RFC #3625)

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2007-09-13  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-12  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-09-12  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-09-11  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-09-11  Tim-Philipp Müller  <tim at centricular dot net>

	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).

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2007-09-10  Wim Taymans  <wim.taymans@gmail.com>

	* 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

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2007-09-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gstdecodebin2.c:
	  Don't leak request pads. Fixes #475395.

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2007-09-09  Sebastian Dröge  <slomo@circular-chaos.org>

	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.

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2007-09-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* tests/check/libs/rtp.c:
	  Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	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).

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-09-06  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Fix configure check for HAVE_LIBXML_HTML.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/.cvsignore:
	  Ignore more, in case the build bots work again one day.

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2007-09-06  Sebastian Dröge  <slomo@circular-chaos.org>

	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.

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2007-09-05  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-09-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Need to include stdlib.h for abs() here too.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c:
	  Fix build.

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2007-09-05  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gststreaminfo.c:
	  Clean up some half-disabled code and comment.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

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	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

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	* 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
870
	event. Fixes #446766.
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	* 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.

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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.

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2007-09-03  Renato Filho  <renato.filho@indt.org.br>
	* gst/playback/gstplaybasebin.c:
	Included "myth://" on stream_uris list for enable buffering to mythtv files

Wim Taymans's avatar
Wim Taymans committed
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2007-09-03  Wim Taymans  <wim.taymans@gmail.com>

	* 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()

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2007-09-03  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-09-03  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* 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()

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2007-08-30  Wim Taymans  <wim.taymans@gmail.com>

	* 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()

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	* 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.

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Daniel Díaz  <yosoy@danieldiaz.org>

	* configure.ac:
	* gst/Makefile.am:
	  Check if libxml provides HTML parser which subparse needs.
	  Fixes #451970.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c:
	  Fix typo and compilation on big endian systems.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstssaparse.c:
	  Convert SSA newline codes into actual newline characters (#470766).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-08-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_plugin_message_get_installer_detail):
	  Add missing separator in PID fallback case.

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2007-08-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-08-23  Sebastian Dröge  <slomo@circular-chaos.org>

1061
	Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
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	* 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.

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2007-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/examples/Makefile.am:
	  Fix even more.

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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* 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

Stefan Kost's avatar
 
Stefan Kost committed
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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  Move passthrough below gst_object_sync_values(). Fixes #442654.

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2007-08-22  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/audio.c:
	Clarify the docs a little.

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2007-08-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  Enable liboil for float and add more details about problems with
	  int16.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
	Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
	Only post buffering messages when we are a stream.

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2007-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.c:
	  Small docs fix and addition.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-textoverlay.c:
	  Add a dumb little test for textoverlay alignments.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	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.

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2007-08-12  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-08-11  Tim-Philipp Müller  <tim at centricular dot net>

	* 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)

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2007-08-10  Stefan Kost  <ensonic@users.sf.net>

	* 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).

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2007-08-10  Michael Smith <msmith@fluendo.com>

	* 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.

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2007-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstqueue2.c:
	* gst/videorate/gstvideorate.c:
	  Printf format fixes (#465028).

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2007-08-09  Michael Smith <msmith@fluendo.com>

	* 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.

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2007-08-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-08-09  Stefan Kost  <ensonic@users.sf.net>

	* ext/ogg/gstoggmux.c:
	  Do not leak oggmux instance.
	
	* ext/vorbis/vorbisenc.c:
	  Also log values.

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2007-08-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/it.po:
	* po/nl.po:
	* po/uk.po:
	* po/vi.po:
	  Updated translations.

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2007-08-08  Stefan Kost  <ensonic@users.sf.net>

	patch by: Yang Hong <hongyang@redflag-linux.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  Add 'silent' property to GstTimeOverlay. Fixes #462979

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2007-08-08  Wim Taymans  <wim.taymans@gmail.com>

1285
	Patch by: Josep Torra Valles <josep@fluendo.com>
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	* 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.

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2007-08-07  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-08-07  Wim Taymans  <wim.taymans@gmail.com>

1305
	Patch by: Josep Torra Valles <josep@fluendo.com>
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	* 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.

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2007-08-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
	Properly respond to conversion queries. Fixes #464079.

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2007-08-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-08-03  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Jens Granseuer <jensgr at gmx dot net>

	* gst/audioconvert/gstaudioquantize.c:
	Fix C89 incompatibilities and spelling of explanations. Fixes #463215.

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2007-08-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
	Add rdt manager for rdt transport.
	Fix parsing of RDT transport.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-08-03  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

Jan Schmidt's avatar
Jan Schmidt committed
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=== release 0.10.14 ===

2007-08-03  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.14, "Light Years Ahead"

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/libs/audio.c: (GST_START_TEST):
	Fix the test to reflect the behaviour of gst_audio_clip_buffer.

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/audio/audio.c:
	When clipping a buffer with no timestamp, assume it is
	within the segment without warnings.

	Fixes: #460978

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
	Fire the signal on the object, not the interface.

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/rtsp/.cvsignore:
	Ber. Don't include the full path, idiot.

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/rtsp/.cvsignore:
	Ignore generated files.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-07-26  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/riff/Makefile.am:
	Pull in our dependencies from -base before those from outside.

Wim Taymans's avatar
Wim Taymans committed
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2007-07-26  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/libs/.cvsignore:
	Ignore the mixer test binary.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	Gratuitous comment change to trigger a rebuild on the buildbots.

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2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

Wim Taymans's avatar
Wim Taymans committed
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2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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Wim Taymans committed
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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
	(vorbis_dec_flush_decode):
	Use the new buffer clipping function from gstaudio here.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

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2007-07-23  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Cleanup the docs.

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2007-07-23  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-07-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gsturidecodebin.c:
	  Init debug category before using it.

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2007-07-21  Jan Schmidt  <thaytan@noraisin.net>

	* 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.

Marc-Andre Lureau's avatar
Marc-Andre Lureau committed
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2007-07-20  Jan Schmidt  <thaytan@noraisin.net>

	* 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

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2007-07-20  Michael Smith <msmith@fluendo.com>

	* 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.

Stefan Kost's avatar
Stefan Kost committed
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2007-07-20  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/volume/Makefile.am:
	* tests/icles/Makefile.am:
	  We should use $(LIBM).

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2007-07-20  Stefan Kost  <ensonic@users.sf.net>

	* tests/icles/Makefile.am:
	  This needs -lm.

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2007-07-16  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	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.

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	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

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-13  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-07-12  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-12  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

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2007-07-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Use pkg-config to locate check.

1833 1834 1835 1836 1837 1838
2007-07-12  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	* tests/check/elements/volume.c: (GST_START_TEST):
	  Fix 'make check' build against core CVS.

1839 1840 1841 1842 1843 1844 1845
2007-07-10  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/propertyprobe.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Make gtk-doc happy.

1846 1847 1848 1849 1850 1851 1852
2007-07-08  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1853 1854 1855 1856 1857
2007-07-06  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Fix location of includes in the docs.

1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869
2007-07-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	(gst_ffmpegcsp_avpicture_fill):
	* gst/ffmpegcolorspace/imgconvert.c: (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

1870 1871 1872 1873 1874 1875
2007-07-05  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	* docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

1876 1877 1878 1879 1880 1881 1882 1883 1884
2007-07-03  Wim Taymans  <wim.taymans@gmail.com>

	* 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.

2007-06-29  Wim Taymans  <wim.taymans@gmail.com>
1885 1886 1887 1888

	* gst/audioconvert/audioconvert.c:
	Include math.h to fix compilation.

1889 1890 1891 1892 1893 1894 1895 1896
2007-06-29  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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

1897 1898 1899 1900 1901 1902 1903 1904 1905 1906 1907 1908 1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919 1920 1921 1922 1923 1924 1925 1926 1927 1928
2007-06-28  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

1929 1930 1931 1932 1933
2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
	Use other metrics as well when estimating the buffer level.

1934 1935 1936 1937 1938 1939 1940 1941 1942 1943 1944
2007-06-28  Wim Taymans  <wim@fluendo.com>

	* 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.

1945 1946 1947 1948 1949 1950 1951 1952 1953 1954
2007-06-28  Wim Taymans  <wim@fluendo.com>

	* 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.

1955 1956 1957 1958 1959 1960 1961 1962 1963 1964 1965
2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1966 1967 1968 1969 1970 1971 1972 1973
2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1974 1975 1976 1977 1978 1979 1980 1981 1982 1983 1984 1985 1986
2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1987 1988 1989 1990 1991 1992
2007-06-27  Wim Taymans  <wim@fluendo.com>

	* 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.

1993 1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008 2009 2010 2011 2012 2013 2014 2015 2016 2017 2018 2019 2020 2021 2022 2023 2024 2025 2026
2007-06-25  Stefan Kost  <ensonic@users.sf.net>

	* 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.

2027 2028 2029 2030 2031
2007-06-25  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Add more files with translatable strings (#450875).

2032 2033 2034 2035 2036 2037 2038 2039
2007-06-23  Edward Hervey  <edward@fluendo.com>

	* 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.

2040 2041 2042 2043 2044
2007-06-22  Jan Schmidt  <thaytan@noraisin.net>

	* MAINTAINERS:
	Updating all the maintainers files

2045 2046 2047 2048 2049 2050 2051
2007-06-21  Stefan Kost  <ensonic@users.sf.net>

	* 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.

2052 2053 2054 2055 2056 2057 2058 2059
2007-06-20  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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>

2060 2061 2062 2063 2064
2007-06-19  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_init): Enable pull-mode operation.

2065 2066 2067 2068 2069 2070
2007-06-19  Michael Smith <msmith@fluendo.com>

	* 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.