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2006-08-21  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gdp.xml:
	* gst/gdp/Makefile.am:
	* tests/check/Makefile.am:
	  Move GDP plugin to -base from -bad.  Closes #347783.

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2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_from_vorbiscomment_buffer):
	  Allow id_data_len == 0 (needed for vorbis comments in Speex files).
	  Also add some checks to make sure we don't memcmp() beyond the end of
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	  vorbiscomment buffer if the ID to check for is larger than the buffer.
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	* tests/check/libs/tag.c: (GST_START_TEST):
	  Some more tests for gst_tag_list_from_vorbiscomment_buffer().

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2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
	(gst_vorbis_enc_set_metadata):
	  Use vorbis comment utility functions from libgsttag
	  instead of re-inventing the wheel (partially fixes #347091).

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2006-08-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix leaks. Wait for state transitions that might happen ASYNC, as well
	as some that won't.

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2006-08-21  Wim Taymans  <wim@fluendo.com>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	Don't try to GObject scan the netbuffer as it's not a GObject.
	Fixes #351308.

	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	* gst-libs/gst/netbuffer/gstnetbuffer.h:
	Document GstNetBuffer.

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2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Add testcase for caps-size-explosion

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2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_get_unit_size), (set_structure_widths):
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	  Lower debug, use g_assert in _get_unit_size
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	* gst/audioresample/gstaudioresample.c:
	(audioresample_get_unit_size):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
	  use g_assert in _get_unit_size

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2006-08-18  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
	(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
	(gst_rtp_buffer_get_payload_buffer):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Document GstRTPBuffer.
	Added function to efficiently strip payload headers.
	API: gst_rtp_buffer_get_payload_subbuffer()

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2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
	(gst_tag_to_vorbis_comments):
	  Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
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	  tags and deserialise them properly as well (#347091).
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	  Add some more gtk-doc blurbs and also some g_return_if_fail().

	* tests/check/libs/tag.c: (GST_START_TEST),
	(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
	  More tests.

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2006-08-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogg.c: (plugin_init):
	* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
	(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
	(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
	(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
	(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
	(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
	Added ogg-in-avi parser element. Fixes #140139.

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
	Fixed a bug in oggdemux debug code.

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Recognise Ogg in the AVI extensible wave format.

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2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  Make buffer durations add up (duration should be next_ts-ts for
	  perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
	  from CVS.

	* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
	(test_buffer_timestamps), (cddabasesrc_suite):
	  Add unit test for the above.

	* tests/check/Makefile.am:
	  Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
	  to see what happens.

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2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
	(gst_alsasink_open):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
	(gst_alsasrc_open):
	Avoid setting and using a NULL device name.
	Print more info when we fail to open a device.

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2006-08-16  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_parse_extended_comment):
	  API: add gst_tag_parse_extended_comment() (#351426).

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
	  Add unit test for gst_tag_parse_extended_comment().

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2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
	  Fix leak (#351502).

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Tim-Philipp Müller committed
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2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* gst/playback/gstplaybin.c:
	  Document playbin.
	  
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.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 to CVS version.

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2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_set_property), (gst_play_bin_get_property),
	(value_list_append_structure_list),
	(gst_play_bin_handle_redirect_message),
	(gst_play_bin_handle_message):
	  Add "connection-speed" property; re-order redirect messages with
	  multiple redirect locations depending on the minimum bitrate if
	  that information is available and a connection speed is set
	  (#350399).

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2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Update max volume to the same value that the volume element uses.

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2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
	Less uglyness..

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2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	(gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
	Add some more debug info.
	Don't crash when a seek failed.
	Actually return the result of the seek instead of TRUE.
	Ignore multiple BOS pages with the same serial so that we don't create
	the same stream multiple times.
	Post an error when we fail to do the initial seek.

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2006-08-13  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
	(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
	Small code cleanup.

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
	(gst_alsa_mixer_new):
	Remove hack that always set the device to hw:0*.
	Properly find the card name for whatever device was configured.
	Do some better debugging.
	Fixes #350784.

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_set_property),
	(gst_alsa_mixer_element_change_state):
	Cleanups.
	Handle setting of a NULL device name better.

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2006-08-11  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c:
	Don't clip float values. Fixes #350900.

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2006-08-11  Andy Wingo  <wingo@pobox.com>

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	* gst/tcp/gsttcp.c: Really fix the build?

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	* gst/tcp/gsttcp.h: For now, always disable deprecation here --
	fixes the build.

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2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
	  Float caps shouldn't have a "signed" field.

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2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
	  Implement SEEKING query in its most basic form, so that we can
	  at least check if we're seekable or not (#350655).

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2006-08-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  The checks here are not even close to anything that would
	  justify MAXIMUM probability, lowering to POSSIBLE until someone
	  fixes the checks (case at hand: quicktime redirection files
	  might start with 00 00 01 XX and pass the checks here just
	  fine, see #350399).

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2006-08-08  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
	  Better detection for multipart/x-mixed-replace: accept leading
	  whitespaces before the boundary marker as well (as our very own
	  multipartmux used to produce) (#349068).

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2006-08-07  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha  <ganadist at chollian net>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	  Detect DTS audio streams (#350157).

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2006-08-05  Andy Wingo  <wingo@pobox.com>

	* ext/theora/gsttheoraparse.h:
	* ext/theora/theoraparse.c (gst_theora_parse_class_init)
	(theora_parse_dispose, theora_parse_set_property)
	(theora_parse_get_property, theora_parse_munge_granulepos)
	(theora_parse_push_buffer, theora_parse_change_state): Add a
	property 'synchronization-points' to fix badly synchronized oggs.

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/audio.c: (structure_contains_channel_positions),
	(fixed_caps_have_channel_positions), (GST_START_TEST),
	(audio_suite), (main):
	  Add a few tests for the channel position stuff in libgstaudio.

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
	(gst_alsa_detect_channels):
	* ext/alsa/gstalsasink.c:
	  Add support for cards that (only) do more than 8 channels,
	  like the Delta 44 (#345188).

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions):
	* gst-libs/gst/audio/multichannel.h:
	  API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
	  unspecified channel position and cannot be combined with any
	  of the other audio channel positions; adjust position layout
	  checks accordingly (#345188).

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Recognise ancient RealAudio files (see #349779).

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jens Granseuer  <jensgr at gmx net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Add typefinder for Interplay's MVE format (#348973).

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2006-08-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Marcel Moreaux <marcelm at luon dot net>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_add_to_queue):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Handle RTP sequence number rollover.
	Disable jitterbuffer by default.

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2006-07-28  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audioresample/gstaudioresample.c: (audioresample_stop),
	(audioresample_set_caps):
	Don't leak references to the incoming caps. Clean them up when
	stopping.

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	(gst_video_scale_finalize):
	Don't leak our temporary pixel buffer.

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
	(GST_START_TEST), (simple_launch_lines_suite):

	Fix leaks and re-enable the test for valgrind checking.

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2006-07-28  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
	(plugin_init):
	  Add typefind function for multipart/x-mixed-replace (#348916).

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2006-07-28  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c: (gst_adder_setcaps),
	(gst_adder_query_duration):
	Fix leak in duration query.
	Reflow some docs and notes.

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2006-07-28  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
	(vorbisenc_suite):
	  Enable Andy's extra vorbisenc test, now that it passes. Also fix one
	  aspect of it.

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2006-07-28  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
	(gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
	(gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	* ext/vorbis/vorbisenc.h:
	  Handle discontinuities in the input vorbis stream correctly,
	  so that the output is properly timestamped (and has good granulepos
	  values). Needs some oggmux fixes too.

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2006-07-27  Wim Taymans  <wim@fluendo.com>

	patch by: Kai Vehmanen <kv2004 eca cx>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_change_state):
	Don't send multiple newsegments with different formats.
	Fixes #348677.

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2006-07-26  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
	Make seeking in ogg more accurate again by doing the more correct
	granuletime to stream time conversion.

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2006-07-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_new_client):
	  

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2006-07-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_new_client):
	  debug a little more understandably
	  do not use goto as a substitute for break, especially if
	  break is also being used

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2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
	* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Remove GLib-2.6 compatibility cruft.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Don't try to align a sample to an unknown value.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
	When the audio clock is slaved to another clock, never try to align
	samples but trust the rate interpolation algorithm.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	Don't try to calculate silence samples, base class does this much
	better now.

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_acquire):
	Calculate silence samples correctly.

	* gst-libs/gst/audio/gstringbuffer.h:
	Add _CAST macro.

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2006-07-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
	  Limit search for the first markup tag to the first few kB of
	  the file. If we don't find one there, it's highly unlikely that
	  this is an XML(-ish) file.

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2006-07-21  Andy Wingo  <wingo@pobox.com>

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	* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
	test to the one in vorbisenc. Also commented out.

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	* tests/check/pipelines/vorbisenc.c: 
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	(test_discontinuity): New test, commented out until Mike lands
	some elite vorbisenc patches.

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	* tests/check/pipelines/theoraenc.c: Port to bufferstraw.
	Bufferstraw was actually factored out of these tests. Now we share
	code yay.

	* configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
	for bufferstraw addition to gstcheck.

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2006-07-21  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (clip_buffer):
	Better clipping.

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2006-07-21  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
	(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
	(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
	Fix leak.
	Avoid type casting when we can.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
	Fix mem leak.

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2006-07-20  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_change_state):
	  Make state change fail if the specified device can't be opened
	  for some reason.

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2006-07-20  Wim Taymans  <wim@fluendo.com>

	* gst/playback/test.c: (gen_video_element), (gen_audio_element),
	(cb_newpad), (main):
	Example of a small audio/video player using decodebin.

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2006-07-20  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
          Add 'fact' chunk id

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2006-07-19  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_change_state):
	Don't assert when not negotiated but post a meaningfull 
	error message. Fixes #347918.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	Add comment about better default MTU size.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	Small cleanups, start docs.

531 532 533 534 535 536 537 538 539
2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Martin Szulecki

	* sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
	  If "device-name" is requested and the device is not
	  open, try to temporarily open it to obtain this
	  information (#342494).

540 541 542 543 544 545 546 547 548
2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstid3tag.c:
	  Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).

	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Some more random const-ifications.

549 550 551 552 553 554 555 556 557
2006-07-18  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c:
	(gst_riff_create_video_template_caps):
          Add more FOURCCs (sort list to make stuff easier to find),
          add comment what those 16 bytes in struct _gst_riff_strh according to
          one avi-dumper are

558 559 560 561 562 563 564
2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions),
	(gst_audio_fixate_channel_positions):
	  Const-ify two arrays.

565 566 567 568 569 570
2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
	  Fix typo, so that alsasink also advertises 8 channels
	  if that's supported (tags: can, worms, open, alsa, ph34r).

571 572 573 574 575 576 577 578
2006-07-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
	*sigh*, when is the compiler going to warn when the comments
	are out-of-sync with the code.. Refix case of busted theora
	headers with 0 granule pos.

579 580 581 582 583 584 585 586 587 588
2006-07-14  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_wait),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	Fix 99% cpu load by waiting for absolute times on the
	clock. Fixes #347300.

589 590
2006-07-14  Andy Wingo  <wingo@pobox.com>

591 592 593 594 595 596 597
	* ext/theora/gsttheoraparse.h: 
	* ext/theora/theoraparse.c (theora_parse_drain_event_queue)
	(theora_parse_push_headers, theora_parse_clear_queue)
	(theora_parse_drain_queue_prematurely, )
	(theora_parse_sink_event, theora_parse_change_state): Queue events
	until we initialized our state, like in vorbisparse.

598 599 600 601 602 603 604 605 606 607 608 609 610 611
	* ext/vorbis/vorbisparse.h: 
	* ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
	(vorbis_parse_push_headers, vorbis_parse_clear_queue)
	(vorbis_parse_drain_queue_prematurely, )
	(vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
	until we have initialized our state. Fixes seeking after an
	initial pad block.

2006-07-14  Andy Wingo  <wingo@pobox.com>

	Patch by: Iain * <iaingnome@gmail.com>
	
	* ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.

612 613 614 615 616
2006-07-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump nano back to CVS

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=== release 0.10.9 ===

2006-07-13  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.9, "I walk the line"

624 625 626 627 628 629 630
2006-07-14  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (stop_pipeline):
	  Move a g_cond_signal to earlier to avoid sometimes deadlocking
	  (commonly happens when running this test under valgrind) when trying
	  to remove the buffer probe.

631 632 633 634 635
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
	Fix missing g_unlock from the previous commit

636 637 638 639 640 641 642 643 644
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_change_state):
	Implement a locking order to ensure we always take the object lock
	before the x_lock and never vice-versa.

645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660
2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (find_compatibles):
	Fix a caps leak when linking (#347304)

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
	(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
	Don't leak shared memory resources. Use the object lock to protect
	against the xcontext disappearing while returning a buffer from the
	pipeline. (#347304)

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
	(vorbis_handle_comment_packet):
	gst_tag_list_merge() returns a new object. Take that into account when
	using it. This avoids memleak.
	Revert previous commit which is not needed.

669 670 671 672 673
2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
	Reset the decoder in finalize so that all fields get cleared.

674 675 676 677 678 679 680 681 682
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_set_clock),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
	Don't try to post an error message when setting the clock fails
	as this can happen when adding an element to a bin which will then
	deadlock. Fixes #347296.

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet):
	Post tag messages on the bus even if we're not initialized.
	If we're not initialized, we still postpone the event pushing of tags.

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2006-07-12  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Revert last two changes that broke the freeze.

698 699 700 701 702
2006-07-12  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	basesink calculates silence sample correctly for us.

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2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Calculate correct silence samples so we don't fill our ringbuffer
	with noise.

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(gst_vorbis_dec_reset), (vorbis_dec_sink_event),
	(vorbis_handle_comment_packet), (vorbis_handle_type_packet):
	* ext/vorbis/vorbisdec.h:
	Delay sending events (newsegment, tags) until the decoder is properly
	initialized.
	Fixes #347295

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2006-07-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audioconvert.c: (get_float_mc_caps),
	(get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
	  Patch from #347221 adding a test for audioconvert
	  channel remappings.

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2006-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
	(gst_ssa_parse_parse_line):
	  Don't include the terminating NUL in the buffer size,
	  it's only there for extra paranoia (would add random
	  '*' characters at the end of each subtitle since the
	  terminator itself is not valid UTF-8 technically).
	  Also fix indenting after boilerplate macro.

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2006-07-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (close_pad_link):
	  Also emit 'unknown-type' signal (which should really be
	  called unhandled-type) if we found potential decoders/demuxers
	  in the registry but none of them worked in the end (as in the
	  case where the plugins don't exist any longer but are still
	  listed in the registry). Fixes #329798.

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2006-07-08  Andy Wingo  <wingo@pobox.com>

	* theoraparse.c (theora_parse_push_buffer)
	(theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
	Add some more debugging. Fix granulepos reconstruction in the face
	of discontinuities.

753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768 769 770 771 772 773
2006-07-06  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init),
	(gst_base_audio_sink_provide_clock):
	Use gobject_class instead of G_OBJECT_CLASS (klass)

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_init),
	(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
	(gst_base_audio_src_get_time),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
	(gst_base_audio_src_create_ringbuffer):
	Fix latency and buffer-time constants and properties ala basesink.
	Implement pull based scheduling. Fixes #346527.
	Set default blocksize in GstBaseSrc to 0, we default to pushing out
	one segment.
	Refuse slaving to another clock instead of silently not working.
	Only provide a clock when we are actually able to do so.
	Various small cleanups and compiler hints.

774 775 776 777 778 779 780 781
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Lutz Mueller <lutz at topfrose de>

	* gst/typefind/gsttypefindfunctions.c: (html_type_find),
	(plugin_init):
	  Add typefinding for text/html (#346581).

782 783 784 785 786 787 788
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	(xml_check_first_element), (xml_type_find), (smil_type_find):
	  Fix SMIL typefinding, make xml_check_first_element() more
	  useful.

789 790 791 792 793 794 795 796 797 798 799 800
2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	(gst_play_base_bin_finalize), (decodebin_element_added_cb),
	(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
	* gst/playback/gstplaybasebin.h:
	  Protect list of elements with a subtitle-encoding property and
	  the subtitle encoding member itself with a lock of their own
	  instead of using the object lock. This prevents a dead-lock in
	  the element-remove callback in some circumstances when shutting
	  down playbin.

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2006-07-05  Sebastien Moutte  <sebastien@moutte.net>

	* win32/common/libgsttag.def:
	Export some new functions.
	* win32/vs6/libgstogg.dsp:
	Add a link to libgsttag-0.10.lib.

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2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Some const-ification.

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2006-07-04  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
	Improve checking if we are dealing with a stream. Added some
	more uris that need buffering.

819 820 821 822 823
2006-07-03  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_do_clip):
	Remove unused variable.

824 825 826 827 828 829 830
2006-07-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	  include lcov.mak
	* configure.ac:
	  add GCOV_LIBS to GST_LIBS

831 832 833 834 835 836 837 838
2006-07-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Michael Sheldon  <webmaster at mikeasoft com>

	* ext/alsa/gstalsasrc.c:
	  Add 32 bps to template caps and increase channels range
	  from [1,2] to [1,MAX]. See #346326.

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2006-06-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Recognise 'WMVA' video codec fourcc (#345879).
	  
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2006-06-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	 
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c: 	 
	  Fixed nasty memory leak

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2006-06-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
	(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
	  fix logging

855 856 857 858 859 860 861 862 863 864 865
2006-06-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
	(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
	Protect remove_fakesink using a mutex, so that we don't try and
	remove the fakesink simultaneously from multiple threads.

	When going from READY to PAUSED, restore the fakesink, so that
	it is there when decodebin gets reused.

866 867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883 884 885
2006-06-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gsttcpclientsink.c:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversink.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/videorate/gstvideorate.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* sys/v4l/gstv4ljpegsrc.c:
	* sys/v4l/gstv4lmjpegsink.c:
	* sys/v4l/gstv4lsrc.c:
	* tests/examples/seek/scrubby.c:
	* tests/examples/seek/seek.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).

886 887 888 889 890 891
2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
	  Second field in GEnumValue shouldn't be a description,
	  but a stringified version of the enum value.

892 893 894 895 896 897 898 899 900
2006-06-22  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
	(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
	Avoid type checking in buffer casts.
	Avoid caps copy in buffer_alloc when we can.
	Use pad_peer_accept.

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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tag.h:
	  Oops, make that 'Since: 0.10.9'.

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2006-06-22  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: (register_tag_image_type_enum),
	(gst_tag_image_type_get_type):
	  API: add GstTagImageType enum to describe images contained
	  in image tags (#345641).

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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	  Fix warnings with gst-inspect: "buffers-min" property
	  should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
	  typo in property description.

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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Cody Russell <bratsche at gnome org>

	* gst/audioresample/gstaudioresample.c:
	(gst_audioresample_class_init):
	* gst/playback/gststreamselector.c:
	(gst_stream_selector_class_init):
	* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
	* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
	* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_class_init):
	* gst/volume/gstvolume.c: (gst_volume_class_init):
	  Avoid unnecessary class cast check in class_init
	  functions (#337747).

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2006-06-21  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
	(gst_text_overlay_video_chain):
	  g_markup_escape_text() REALLY doesn't like non-UTF8 input
	  and doesn't validate its input either (and neither did
	  textoverlay it seems). Let's do that then and fix #345206.

951 952 953 954 955 956 957 958 959 960 961 962 963 964 965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980
2006-06-19  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
	(gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
	(find_syncframe), (find_limits), (assign_value),
	(count_burst_unit), (gst_multi_fd_sink_new_client),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
	(gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
	(gst_multi_fd_sink_change_state):
	* gst/tcp/gstmultifdsink.h:
	Added shiny new burst-on-connect methods.
	Add properties to control the minimal amount of data queued.
	Small cleanups.
	API: bytes-min property
	API: time-min property
	API: buffers-min property
	API: burst-unit property
	API: burst-value property
	API: add-full signal

	* gst/tcp/gsttcp-marshal.list:
	Added new marshaller code for the new signal.

	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(multifdsink_suite):
	Added testcases for new burst methods.

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2006-06-19  Edward Hervey  <edward@fluendo.com>

	* ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
	Implement clipping for accurate seeking.
	Closes #345225

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2006-06-19  Wim Taymans  <wim@fluendo.com>

	Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
	(gst_video_scale_transform):
	Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131

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2006-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
998
	  Fix --disable-external (can't set conditionals conditionally,
999 1000
	  #343602).

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/audioresample.c: (test_reuse),
	(audioresample_suite):
	  Add test case for bug #342789 fixed below.

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioresample/gstaudioresample.c:
	(gst_audioresample_class_init), (gst_audioresample_init),
	(audioresample_start), (audioresample_stop),
	(gst_audioresample_set_property), (gst_audioresample_get_property):
	  Implement GstBaseTransform::start and ::stop so that audioresample
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	  can clear its internal state properly and be reused instead of
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	  causing non-negotiated errors with playbin under some circumstances
	  (#342789).

	* tests/check/elements/audioresample.c: (setup_audioresample),
	(cleanup_audioresample):
	  Need to set element state here so that ::start and ::stop are
	  called.

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2006-06-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Young-Ho Cha <ganadist at chollian dot net>

	* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
	Parse extra data better, apparently it's right behind
	the normal strf header size. Fixes #343500.

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2006-06-16  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams):
	If we fail to set the buffer_time and period_time alsa
	parameters, post a warning and leave alsa select a 
	default instead of failing. Fixes #342085

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/cdda/gstcddabasesrc.h:
	  Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
	  out in the header file and shouldn't be listed in the docs.

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
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	  Must dereference pointer to fourcc in the debug statement.
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2006-06-16  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	add remaining symbols into correct setions
	
	* gst-libs/gst/audio/gstringbuffer.c:
	fix incomplete docs
	
	* gst-libs/gst/audio/gstringbuffer.h:
	comment out not yet implemented function
	
	
	* gst-libs/gst/floatcast/floatcast.h:
	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	add short descriptions
	
	
	* gst-libs/gst/interfaces/propertyprobe.c:
	fix return value docs	
	
	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	simplify debug logging
	
	* gst-libs/gst/riff/riff-read.h:
	sync function prototype and docs
	
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	remove left over symbol

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* autogen.sh:
	* configure.ac:
	* docs/Makefile.am:
	  Use GST_PLUGIN_DOCS macro in configure.ac, add
	  --enable-plugin-docs default to autogen.sh and use
	  ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).

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2006-06-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
	(gst_ogg_demux_loop):
	Combine GstFlowReturn from the source pads to give a
	meaningfull result to the upstream peer or to stop the
	processing task in case of errors.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c: (cb_probe):
	  Try GST_TAG_CODEC as fallback when extracting the
	  codec name; more debug info.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Extract language tags from ogm subtitle streams, so that
	  the subtitle menu choices are labelled correctly in
	  Totem (fixes #344708).

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2006-06-14  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
	(gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
	(gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
	(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
	Fix various leaks. Fixes #343699.
	Add x-smoke mime type.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-ids.h:
	  Add IDs for 'bext' chunks (see #343837).

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2006-06-12  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha  <ganadist at chollian net>

	* gst/subparse/samiparse.c: (sami_context_pop_state),
	(handle_start_font), (end_sami_element):
	  Honour font face tags in SAMI subtitles (#344503).

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* po/POTFILES.in:
	  add missing files containing translatable strings

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/tmpl/.cvsignore:
	  we don't want those *.sgml files in CVS either

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/.cvsignore:
	* tests/check/elements/.cvsignore:
	* tests/check/libs/.cvsignore:
	  ignore more

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	  also commiting the changed Makefile.am (added more libs to the
	  doc-build)

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	  first batch of reordering things, add index & hierarchy

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2006-06-11  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  use GST_PKG_CHECK_MODULES, cleans up output

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2006-06-10  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
	  Add support for burn:// URIs (#343385); const-ify things a bit,
	  use G_N_ELEMENTS instead of hard-coded array size.

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2006-06-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha  <ganadist at chollian net>

	* gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
	  Fix up broken entities before passing them to libxml *sigh*.
	  (#343303).
	  
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2006-06-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  back to TRUNK

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=== release 0.10.8 ===

2006-06-09  Thomas Vander Stichele <thomas at apestaart dot org>

	* configure.ac:
	  releasing 0.10.8, "Moar gij ziet mij nie"

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2006-06-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* win32/common/config.h:
	  0.10.7.2 prerelease

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2006-06-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/libs/tmpl/gstaudio.sgml:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* docs/libs/tmpl/gsttuner.sgml:
	* docs/libs/tmpl/gstxoverlay.sgml:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/gstringbuffer.c:
	* gst-libs/gst/interfaces/colorbalance.c:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/interfaces/tuner.c:
	* gst-libs/gst/interfaces/xoverlay.c:
	  move last template doc snippets to source code and delete them

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2006-06-06  Michael Smith  <msmith@fluendo.com>

	* ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
	(theora_parse_drain_queue):
	  Mark DELTA_UNIT on non-keyframes.

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2006-06-03  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_samples_done):
	* gst-libs/gst/audio/gstringbuffer.h:
	Document better the fact that latency_time and buffer_time are values
	stored in microseconds, and not the usual GStreamer nanoseconds.
	Change the variables (compatibly) that store them from GstClockTime 
	to guint64 to make it more clear that they're not storing clock times.
	Also, remove the bogus property description that says the user can
	specify -1 to get the default value, since that's never been the case.

	When computing the default segment size for the ring buffer, make it
	an integer number of samples.

	When the sub-class indicates a delay greater than the number of
	samples we've written return 0 from the audio sink get_time method.

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* tests/check/elements/audioconvert.c: (set_channel_positions),
	(get_float_mc_caps), (get_int_mc_caps):
	* tests/check/elements/audioresample.c:
	* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
	* tests/check/elements/videorate.c:
	* tests/check/elements/videotestsrc.c: (GST_START_TEST):
	* tests/check/elements/volume.c:
	* tests/check/elements/vorbisdec.c:
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
	  Don't busy-wait in tests; this was causing test timeouts very
	  frequently when running under valgrind.

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2006-06-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/README:
	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
	(gst_multi_fd_sink_remove_client_link),
	(gst_multi_fd_sink_client_queue_caps),
	(gst_multi_fd_sink_client_queue_buffer),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_render):
	* gst/tcp/gstmultifdsink.h:
	  make multifdsink properly deal with streamheader:
	  - streamheader is taken from caps
	  - buffers marked with IN_CAPS are not sent
	  - streamheaders are sent, on connection, from the caps of the
	    buffer where the client gets positioned to
	  - further streamheader changes are done every time the client
	    will receive a buffer with different caps
	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(gst_multifdsink_create_streamheader):
	  add tests for this

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	  Reinstate limit on channel count. Vorbis does not define the meaning
	  of > 6 channels, so they're just independent channels. Gstreamer
	  currently has no mechanism to represent N independent channels.

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	  Don't arbitrarily restrict channel counts and rate in vorbis.
	  In terms of effects likely on real-world files, this fixes 96kHz
	  playback of vorbis.

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/audioconvert.c: (float):
	  More correct float->int conversion.

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
	  Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
	  value. Fixes g-critical on trying to play back ogg containing
	  unknown codec.

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2006-06-02  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (group_create), (group_commit),
	(setup_source):
	* gst/playback/gstplaybasebin.h:
	Make the subtitle detection work from any thread so we don't
	deadlock. Fixes #343397.

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2006-06-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/volume/Makefile.am:
	  Seriously, it's not *that* hard to get compilation right.  Even
	  a drunk can do it ! Add LIBOIL CFLAGS and LIBS

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2006-06-01  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_class_init),
	(gst_volume_init), (volume_process_float), (volume_process_int16),
	(volume_process_int16_clamp), (volume_set_caps),
	(volume_transform_ip), (plugin_init):
	* gst/volume/gstvolume.h:
	rewrite the passthrough check, split _int16 and _int16_clamp, fix
	another property desc., remove unused param from process function
	
	* tests/check/elements/volume.c: (volume_suite):
	reactivate the passthrough test

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2006-06-01  Stefan Kost  <ensonic@users.sf.net>

	* ext/alsa/gstalsamixerelement.h:
	* ext/alsa/gstalsamixeroptions.h:
	* ext/alsa/gstalsamixertrack.h:
	* ext/gnomevfs/gstgnomevfssink.h:
	* ext/gnomevfs/gstgnomevfssrc.h:
	* ext/theora/gsttheoradec.h:
	* ext/theora/gsttheoraenc.h:
	* ext/theora/gsttheoraparse.h:
	* ext/vorbis/vorbisparse.h:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioresample/gstaudioresample.h:
	* gst/audiotestsrc/gstaudiotestsrc.h:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
	* gst/playback/gststreamselector.h:
	* gst/tcp/gstmultifdsink.h:
	* gst/tcp/gsttcpclientsink.h:
	* gst/tcp/gsttcpclientsrc.h:
	* gst/tcp/gsttcpserversink.h:
	* gst/tcp/gsttcpserversrc.h:
	* gst/videorate/gstvideorate.h:
	* gst/videoscale/gstvideoscale.h:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/volume/gstvolume.h:
	* sys/v4l/gstv4ljpegsrc.h:
	* sys/v4l/gstv4lmjpegsink.h:
	* sys/v4l/gstv4lmjpegsrc.h:
	* sys/v4l/gstv4lsrc.h:
	* sys/ximage/ximagesink.h:
	* sys/xvimage/xvimagesink.h:
	* tests/old/testsuite/alsa/sinesrc.h:
	Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

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2006-05-31  Wim Taymans  <wim@fluendo.com>

	* ext/libvisual/visual.c: (gst_visual_reset),
	(gst_visual_sink_setcaps), (gst_visual_sink_event),
	(gst_visual_src_event), (get_buffer), (gst_visual_chain):
	Handle DISCONT.
	Use running time before doing QoS.
	Handle mono too.

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2006-05-31  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/libs/Makefile.am:
	  set a magic variable to indicate we know the docs are incomplete

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2006-05-30  Sebastien Moutte  <sebastien@moutte.net>

	* win32/common/libgstvideo.def:
	  export gst_video_calculate_display_ratio
	* win32/vs6/libgstvideoscale.dsp:
	  add link to libgstvideo-0.10.lib

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2006-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gen_source_element):
	  Throw a more comprehensible error for rtsp:// URIs (rather
	  than erroring out with a negotiation error later on) until
	  we fix playbin to handle rtspsrc etc.

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2006-05-30  Wim Taymans  <wim@fluendo.com>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
	(gst_text_overlay_text_event):
	Added some FIXMEs.

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2006-05-30  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
	(gst_adder_request_new_pad), (gst_adder_release_pad):
	* gst/adder/gstadder.h:
	Implement release_request_pad.
	Make padcounter atomic.

	* tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
	Added check for release_pad in adder.

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2006-05-30  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
	Fix build again.

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2006-05-30  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
	(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
	(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	(gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
	(gst_ogg_demux_seek), (gst_ogg_demux_get_data),
	(gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
	(gst_ogg_demux_bisect_forward_serialno),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
	(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
	  add more debugging
	  clean up printf formats for granulepos and serialno

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2006-05-30  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (raw_caps_factory),
	(gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
	(gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
	(gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
	(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	* ext/vorbis/vorbisenc.h:
	  Multi-channel caps negotiation, so we can do proper multichannel
	  vorbis encoding, negotiated through audioconvert.

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2006-05-30  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/adder.c: (test_event_message_received),
	(test_play_twice_message_received), (GST_START_TEST),
	(adder_suite):
	Added check to show that #339935 is fixed with ongoing
	adder and collectpads fixes.

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2006-05-29  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c: (gst_adder_request_new_pad):
	Don't leak pad name.

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2006-05-29  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c: (gst_adder_query_duration),
	(forward_event_func), (forward_event), (gst_adder_src_event):
	Fix adder seeking.
	Make query/seeking code threadsafe.

	* tests/check/Makefile.am:
	* tests/check/elements/adder.c: (test_event_message_received),
	(GST_START_TEST), (test_play_twice_message_received):
	Fix adder test case.

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2006-05-29  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Young-Ho Cha  <ganadist at chollian net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (gst_play_base_bin_dispose),
	(set_encoding_element), (decodebin_element_added_cb),
	(decodebin_element_removed_cb), (setup_subtitle), (setup_source),
	(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
	* gst/playback/gstplaybasebin.h:
	  Add 'subtitle-encoding' property to playbin, so applications can
	  force a subtitle encoding for non-UTF8 subtitles (#342268).

	* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
	(gst_sub_parse_set_property):
	  Rename recently-added 'encoding' property to 'subtitle-encoding'
	  (so it can be proxied by playbin/decodebin in a generic way
	  with less danger of false positives).

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2006-05-29  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(append_with_other_format), (set_structure_widths),
	(gst_audio_convert_transform_caps):
	  Patch from #341562: give more specific audio caps in get_caps, so
	  that basetransform  can make better decisions on what caps to
	  negotiate.

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2006-05-28  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/volume.c:
	make it compile again

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2006-05-28  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/volume.c: (volume_suite):
	disable test until #343196 gets resolved

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2006-05-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c: (gst_adder_get_type):
	Make it easier to copy&paste
	
	* gst/volume/Makefile.am:
	* gst/volume/gstvolume.c: (volume_update_real_volume),
	(gst_volume_set_volume), (gst_volume_set_mute),
	(gst_volume_class_init), (volume_process_int16), (volume_set_caps),
	(volume_transform_ip), (volume_update_mute),
	(volume_update_volume):
	* gst/volume/gstvolume.h:
	Add own debug category, move duplicate code to helper function, fix
	property texts, add more comments and prepare ffor liboil-goodness
	
	* tests/check/Makefile.am:
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	add test for mute and passtrough case, be a bit more verbose to track
	failure
	
	* tests/check/generic/states.c: (GST_START_TEST):
	catch elements that fail to instantiate

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2006-05-28  Edward Hervey  <edward@fluendo.com>

	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/theoraenc.c:
	* tests/check/pipelines/vorbisenc.c:
	Comment out tests using parse_launch() if core was built without
	parsing capabilities.

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2006-05-27  Edward Hervey  <edward@fluendo.com>

	* tests/check/Makefile.am:
	Extra bonus points for whoever explains to ensonic that you are meant
	to test unit tests thoroughly before commiting them, especially if
	you know it's going to break.
	De-activated element/adder tests.

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2006-05-27  Edward Hervey  <edward@fluendo.com>

	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
	Marking caps conversion issues as GST_WARNING is way too verbose,
	Moving them to GST_LOG.

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2006-05-27  Tim-Philipp Müller  <tim at centricular dot net>

	* README:
	  Replace current README (containing the release notes from
	  some 0.9.x version) with a proper README taken from the core.

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2006-05-26  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	(vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
	(vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
	(vorbis_dec_change_state):
	Small cleanups.
	Add some FIXMEs
	Clip output samples to segment boundaries.

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2006-05-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
	Improve the errors produced on bad output, including some human
	readable description strings.
	Handle the (theoretical for ximagesink) case where the XServer 
	has a different idea about the size required for a particular 
	frame and gives us too small a memory allocation.

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2006-05-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
	(gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
	Improve the errors produced on bad output, including some human 
	readable description strings.
	Handle RGB Xv formats properly by transforming them into our 
	big-endian caps description.
	Use gst_caps_truncate to ensure that we never try and choose a
	non-fixed caps in buffer_alloc.
	Handle the case where the XServer has a different idea about the size
	required for a particular frame and gives us too small a memory
	allocation.
	Use -1 to indicate 'no image format', because 0 is a valid XServer
	image format number.
	Put RGB Xv formats at the end of the caps, so that we always prefer
	YUV format frames.
	Iterate the available Xv Encodings to determine the maximum width and
	height, and then return that in our caps.
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	(Closes #315312, #337544)
1618

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2006-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
	When there is only one unfinished pad and it receives an event that
	doesn't match our requirements, we need to set alldone=FALSE so that
	the fakesink is not removed yet.

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2006-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
	  Use gst_type_find_helper_for_buffer() to find the type
	  of stream from the first packet.

	* configure.ac:
	  Bump requirements to core CVS (needed for vorbis
	  typefinding to work).

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2006-05-24  Edward Hervey  <edward@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (qt_type_find): 
Thomas Vander Stichele's avatar
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	Added the 'prfl' atom type which MQV (no, it's not a typo) files
	contain.  Else they play perfectly fine with qtdemux.
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2006-05-23  Stefan Kost  <ensonic@users.sf.net>

	* ext/theora/theoradec.c:
	* ext/theora/theoraenc.c:
	* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
	* gst/audiorate/gstaudiorate.c:
	  make more debug catagories static
	
	* tests/check/Makefile.am:
	* tests/check/elements/adder.c: (message_received),
	(test_event_message_received), (GST_START_TEST),
	(test_play_twice_message_received), (adder_suite):
	  added test case for using element twice, extra bonus points for anyone
	  who can make these test run reliably

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2006-05-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/theora/theoradec.c: (theora_dec_chain):
	  Make work with time-stamped input buffers that do not
	  have a granulepos in BUFFER_OFFSET_END (like theora
	  buffers coming from matroskademux). Fixes #342448.

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2006-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/tcp/Makefile.am:
	  fdstresstest doesn't need Gtk+, fix compilation if 
	  gtk is not available (#342566).

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2006-05-19 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
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Thomas Vander Stichele committed
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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c: 	 
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	  80 line columns
	  Removed redundant floor()
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2006-05-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  On second thought, just skip JUNK chunks automatically, so
	  the caller doesn't have to handle this. Fixes #342345.
	  Also, return GST_FLOW_UNEXPECTED if we get a short read,
	  not GST_FLOW_ERROR.

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2006-05-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  Don't bail out on JUNK chunks with a size of 0 (would try to
	  pull_range 0 bytes before, which sources don't like too much).
	  See #342345.

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2006-05-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
	Use the gstutil scaling function to preserve 64 bits while calculating
	output width and height from the display-aspect-ratio. (A continuation
	of #341542)

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2006-05-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
	(gst_xvimagesink_buffer_alloc):
	* sys/xvimage/xvimagesink.h:
	When performing buffer allocations, remember the caps and image format
	we return so that if the same caps are asked for next time we can
	return them immediately without doing any caps intersections.

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2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
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	* gst-libs/gst/rtp/README:
	  Some new documentation
	* gst-libs/gst/rtp/gstrtpbuffer.h: 	 
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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	  Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children 	 
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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c: 	 
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h: 	 
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	  New RTP audio base payloader class. Supports frame or sample based
	  codecs.  Not enabled in Makefile.am until approved.
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2006-05-18  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/alsa.c: (test_device_property_probe):
	  Fix test case: don't try to free NULL GValueArray when there
	  are no devices.

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2006-05-18  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/alsa.c: (test_device_property_probe),
	(alsa_suite), (main):
	  Add simple test that runs a device property probe on alsasrc,
	  alsasink and alsamixer. Disable valgrind check for now (too
	  many leaks in libasound, and valgrind ignored my suppressions
	  additions).

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