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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>

180
	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>

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	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.

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

	* configure.ac:
	Back to CVS

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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.

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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.

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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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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.

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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.

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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.

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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.

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2007-07-06  Stefan Kost  <ensonic@users.sf.net>

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

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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

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2007-07-05  Stefan Kost  <ensonic@users.sf.net>

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

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

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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

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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.

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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.

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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.

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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.

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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.

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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.

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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.

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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.

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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.

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2007-06-25  Tim-Philipp Müller  <tim at centricular dot net>

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

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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.

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2007-06-22  Jan Schmidt  <thaytan@noraisin.net>

	* MAINTAINERS:
	Updating all the maintainers files

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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.

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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>

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2007-06-19  Andy Wingo  <wingo@pobox.com>

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

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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.

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2007-06-17  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/vi.po:
	  Update translations.

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2007-06-15  David Schleef  <ds@schleef.org>

	* gst/playback/gstqueue2.c:
	  Fix compile error from ignored return value.

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2007-06-15  Michael Smith <msmith@fluendo.com>

	* 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.

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2007-06-15  Michael Smith <msmith@fluendo.com>

	* 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.

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2007-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-06-13  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_create_read):
	Fix build on MacOSX.

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2007-06-13  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-06-12  Wim Taymans  <wim@fluendo.com>

	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.

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2007-06-11  Wim Taymans  <wim@fluendo.com>

	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.

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2007-06-07  Sebastien Moutte  <sebastien@moutte.net>

	* 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.

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2007-06-07  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-06-07  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_get_range):
	Fix compilation.

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2007-06-06  Wim Taymans  <wim@fluendo.com>

	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.

Wim Taymans's avatar
Wim Taymans committed
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2007-06-06  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	* gst/playback/gstqueue2.c: (gst_queue_create_read):
	Some more fseeko checks.

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2007-06-06  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	check for large file support.

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2007-06-05  Sebastian Dröge  <slomo@circular-chaos.org>

	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.

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2007-06-01  Michael Smith <msmith@fluendo.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	  Don't overflow intermediate values when seeking to large time values
	  in audiotestsrc.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	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.

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2007-06-05  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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. 

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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:
1180
	Add support for remove_flush.
1181

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* 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.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	  Back to CVS

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=== release 0.10.13 ===

2007-06-05  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.13, "What's Going on?"

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2007-05-31  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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>

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2007-05-31  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/floatcast/floatcast.h:
	Define inline when needed on win32 builds. Fixes: #441295
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	Patch by: Sebastien Moutte  <sebastien@moutte.net>
1217

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2007-05-29  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (queue_overrun),
	(no_more_pads_full):
	Stop buffering when the group is commited because the queues filled up.
	Fixes #442024.

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2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-05-24  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-05-24  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-05-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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

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2007-05-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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

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2007-05-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	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.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cdparanoia/gstcdparanoiasrc.c:
	  Repair umlaut.

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2007-05-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-05-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

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2007-05-21  Michael Smith <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_data_packet):
	  Don't look inside 0-length packets (which indicate duplicated
	  frames)

Wim Taymans's avatar
Wim Taymans committed
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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* 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.

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2007-05-18  Stefan Kost  <ensonic@users.sf.net>

	* 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.

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2007-05-18  Stefan Kost  <ensonic@users.sf.net>

	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).

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2007-05-17  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

1412 1413 1414 1415 1416 1417 1418 1419 1420
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* 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.

1421 1422 1423 1424 1425 1426 1427 1428
2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1429 1430 1431 1432 1433 1434 1435 1436 1437 1438 1439 1440 1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452 1453 1454 1455
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* 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.

1456 1457 1458 1459 1460 1461 1462
2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1463 1464 1465 1466 1467 1468 1469 1470 1471 1472 1473 1474
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* 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.

1475 1476 1477 1478 1479 1480 1481 1482 1483 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493 1494
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* 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.

1495 1496 1497 1498 1499 1500 1501
2007-05-17  Michael Smith <msmith@fluendo.com>

	* 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.

1502 1503 1504 1505 1506
2007-05-15  David Schleef  <ds@schleef.org>

	* docs/Makefile.am: Install docs even when --disable-gtk-doc
	  is disabled.  This matches the behavior of gtk+.  Fixes #349099.

1507 1508 1509 1510 1511 1512
2007-05-15  Wim Taymans  <wim@fluendo.com>

	* 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.

1513 1514 1515 1516 1517 1518 1519 1520 1521
2007-05-15  Wim Taymans  <wim@fluendo.com>

	* 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.

1522 1523 1524 1525 1526
2007-05-15  Michael Smith <msmith@fluendo.com>

	* configure.ac:
	  We don't require a C++ compiler. So don't require one.

1527 1528 1529 1530 1531 1532 1533 1534 1535 1536
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* 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.

1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552 1553 1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564 1565 1566
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	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

1567 1568 1569 1570 1571 1572
2007-05-14  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add support for video/x-raw-bayer.

1573 1574 1575 1576 1577 1578
2007-05-12  David Schleef  <ds@schleef.org>

	* sys/xvimage/xvimagesink.c:
	  Add some sanity checking for the XVImage size returned by X.
	  Related to #377400.

1579 1580 1581 1582 1583 1584 1585 1586
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* 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.

1587 1588 1589 1590 1591 1592 1593
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* 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.

1594 1595 1596 1597 1598 1599 1600 1601 1602 1603 1604 1605 1606 1607 1608 1609 1610 1611 1612 1613 1614
2007-05-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* 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.

1615 1616 1617 1618 1619 1620 1621
2007-05-10  Wim Taymans  <wim@fluendo.com>

	* 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.

1622 1623 1624 1625 1626 1627 1628 1629 1630 1631
2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* 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.

1632 1633 1634 1635 1636
2007-05-09  Michael Smith <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  gst_pad_push_event() does not return a GstFlowReturn!

1637 1638 1639 1640 1641 1642
2007-05-09  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/scrubby.c: (stop_cb), (main):
	* tests/examples/seek/seek.c: (do_seek):
	Some small cosmetic changes.

1643 1644 1645 1646 1647 1648 1649 1650
2007-05-08  Stefan Kost  <ensonic@users.sf.net>

	* 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.

1651 1652 1653 1654 1655 1656 1657 1658 1659 1660
2007-05-07  Michael Smith <msmith@fluendo.com>

	* 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.

1661 1662 1663 1664 1665 1666 1667 1668 1669 1670 1671 1672 1673 1674
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* 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.

1675 1676 1677 1678 1679 1680 1681
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1682 1683 1684 1685 1686 1687 1688
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1689 1690 1691 1692 1693 1694
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix build when disabling asserts.

1695 1696 1697 1698 1699 1700 1701 1702 1703 1704 1705 1706 1707
2007-05-03  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

1708 1709 1710 1711 1712 1713
2007-05-03  Michael Smith <msmith@fluendo.com>

	* 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.

1714 1715 1716 1717 1718 1719
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* 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.

1720 1721 1722 1723 1724 1725
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* 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.

1726 1727 1728 1729 1730 1731
2007-05-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
	* gst/playback/gstdecodebin2.c: (plugin_init):
	  Better error message for text files.

1732 1733 1734 1735 1736
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
	Fix offset bug in generation RR packets.

1737 1738 1739 1740 1741 1742 1743 1744 1745
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.

1746 1747 1748 1749 1750 1751 1752 1753 1754 1755 1756
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* 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.

1757 1758 1759 1760 1761 1762 1763 1764 1765
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* 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).

1766 1767 1768 1769 1770 1771 1772 1773 1774 1775 1776 1777 1778 1779 1780 1781 1782
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* 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.

1783 1784 1785 1786 1787 1788 1789
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>

	* sys/ximage/ximagesink.c:
	  Fix build if XShm is not available (#432362).

1790 1791 1792 1793 1794 1795 1796
2007-04-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

1797 1798 1799 1800 1801 1802 1803 1804 1805 1806 1807 1808
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	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.

1809 1810 1811 1812 1813 1814
2007-04-23  Stefan Kost  <ensonic@users.sf.net>

	* 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

1815 1816 1817 1818 1819 1820 1821 1822 1823 1824 1825
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	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).

1826 1827 1828 1829 1830 1831 1832 1833 1834
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1835 1836 1837 1838 1839 1840 1841 1842 1843 1844 1845 1846
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	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).

1847 1848 1849 1850 1851
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioresample/gstaudioresample.c:
	  Make more functions static, just because we can.

1852 1853 1854 1855 1856
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/audioresample.c:
	  Add unit test for audioresample shutdown crasher (#420106).

1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869 1870 1871 1872
2007-04-20  Stefan Kost  <ensonic@users.sf.net>

	* 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

1873 1874 1875 1876 1877 1878 1879 1880
2007-04-19  Michael Smith  <msmith@fluendo.com>

	* 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).

1881 1882 1883 1884 1885
2007-04-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/.cvsignore:
	Fix build.

1886 1887 1888 1889 1890
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

1891 1892 1893 1894 1895 1896
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.

1897 1898 1899 1900 1901 1902 1903 1904 1905 1906 1907 1908 1909 1910 1911 1912 1913 1914
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* 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.	

1915 1916 1917 1918 1919 1920 1921 1922 1923 1924 1925
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	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.

1926 1927 1928 1929 1930 1931 1932
2007-04-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* 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.

1933 1934 1935 1936 1937 1938 1939
2007-04-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* 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.

1940 1941 1942 1943 1944 1945 1946
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* 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

1947 1948 1949 1950 1951 1952 1953
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* 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.

1954 1955 1956 1957 1958
2007-04-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/theora/theoradec.c: (theora_dec_sink_event):
	Fix misleading comment.

1959 1960 1961 1962 1963
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  More sanity checks for the header fields.

1964 1965 1966 1967 1968 1969
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1970 1971 1972 1973 1974 1975 1976 1977 1978 1979
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* 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.

1980 1981 1982 1983 1984 1985
2007-04-12  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  Use scale functions to avoid overflow when calculating duration of 
	  vorbis buffers.

1986 1987 1988 1989 1990 1991 1992 1993 1994 1995
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.

1996 1997 1998 1999 2000 2001 2002 2003 2004
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* 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.

2005 2006 2007 2008 2009 2010 2011 2012 2013
2007-04-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	* 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

2014 2015 2016 2017 2018 2019
2007-04-06  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybin.c: (add_sink),
	(gst_play_bin_change_state):
	Activate sync in playbin, we are ready to handle it for live streams.

2020 2021 2022 2023 2024 2025
2007-04-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/playbin.c:
	(test_sink_usage_video_only_stream), (playbin_suite):
	  Add small test for stream-info-value-array code paths.

2026 2027 2028 2029 2030 2031 2032
2007-04-05  Wim Taymans  <wim@fluendo.com>

	* 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.

2033 2034 2035 2036 2037 2038 2039 2040
2007-04-05  Wim Taymans  <wim@fluendo.com>

	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.

2041 2042 2043 2044 2045 2046 2047 2048 2049 2050 2051 2052
2007-04-03  David Schleef  <ds@schleef.org>

	* 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.

2053 2054 2055 2056 2057 2058 2059 2060 2061 2062 2063 2064
2007-04-03  Wim Taymans  <wim@fluendo.com>

	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