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2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_pattern_get_type),
	(gst_video_test_src_set_pattern):
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
	(gst_video_test_src_black), (gst_video_test_src_white),
	(gst_video_test_src_red), (gst_video_test_src_green),
	(gst_video_test_src_blue):
	* gst/videotestsrc/videotestsrc.h:
	  Add more uni-colour patterns ("white", "red", "green", and "blue").

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2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
	  Fix stride for YVYU, should be word-aligned (#353658).

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2006-08-31  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/adder/gstadder.c: (gst_adder_src_event):
	  Fix build.

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2006-08-31  Edward Hervey  <edward@fluendo.com>

	* gst/adder/gstadder.c: (forward_event_func),
	(gst_adder_src_event), (gst_adder_collected),
	(gst_adder_change_state):
	* gst/adder/gstadder.h:
	Remember the start position asked in the incoming seeks, so we can
	output GST_EVENT_NEW_SEGMENT with a correct position value (instead
	of assuming it will always be 0).

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2006-08-31  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
	(gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_loop):
	Send the GST_EVENT_NEW_SEGMENT from the streaming thread.

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2006-08-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	  Return FALSE instead of returning a random false unit
	  size when the format isn't known/supported (even if
	  this shouldn't happen under normal circumstances).

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2006-08-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
	(gst_gnome_vfs_src_start):
	Try harder to get the size from a uri by using _info_uri() when
	_info_from_handle() does not give us enough info. 
	Also follow symlinks when getting the size.
	Partially Fixes #332864.

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2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Viktor Peters  <viktor dot peters at gmail dot com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
	(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
	(gst_alsa_mixer_set_record):
	* ext/alsa/gstalsamixertrack.c:
	(gst_alsa_mixer_track_update_alsa_capabilities),
	(alsa_track_has_cap), (gst_alsa_mixer_track_new),
	(gst_alsa_mixer_track_update):
	* ext/alsa/gstalsamixertrack.h:
	  Improve and fix mixer track handling, in particular better handling
	  of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
	  track objects for tracks that have both capture and playback volume
	  (and label them differently as well so they're not mistakenly
	  assumed to be duplicates); classify mixer tracks that only affect
	  the audible volume of something (rather than the capture volume)
	  as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
	  for capture tracks to correspond to alsa-pswitch alsa-cswitch
	  (following the meaning documented in the mixer interface header
	  file); add support for alsa's exclusive cswitch groups; update/sync
	  state/flags better if mixer settings are changed by another
	  application. Fixes #336075.

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2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: add section about BUFFERING messages sent by playbin.

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2006-08-29  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain):
	  Ignore explicit DISCONT marked on buffers (which is often spurious,
	  particularly when using multiple segments), in favour of solely
	  using the timestamps/durations.

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2006-08-29  Edward Hervey  <edward@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	Don't rely on incoming buffers offset anymore, since it is completely
	broken when using multiple segments.
	Instead convert the incoming buffers timestamp to running time, and
	then convert that value to the offsets.
	Also inform GstSegment of the last outputted stop position, which is
	needed if we received several segments with an unknown stop value.

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2006-08-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  fix buffer unreffing on a header push failure

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2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
	(gst_audio_rate_chain):
	Make the metadata of the buffer writable before changing its
	flags.

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2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	(gst_audio_rate_setcaps), (gst_audio_rate_init),
	(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
	(gst_audio_rate_chain), (gst_audio_rate_change_state):
	Fix audiorate some more.
	Reset and resync counters on flush and READY.
	Handle the DISCONT flag correctly.
	Use GstSegment to track position.
	Fail when not negotiated.
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	Fixes #353234.
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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Fix spelling.
	  Remove accidently included debug line.

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2006-08-25  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Small cleanups.
	If a buffer is received with no caps, make the buffer metadata
	writable and set the caps, making sure that we don't screw up the
	refcounts.

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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
	(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
	  Fix memory leaks and misleading debug messages, add a couple of
	  comments.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
	(gst_multi_fd_sink_render):
	  Do not use gst_buffer_make_writable() in a basesink render method,
	  as it may incorrectly unref the buffer. Instead, use convoluted
	  dance to avoid copying the buffer except when we need to.

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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c:
	(gst_vorbis_enc_buffer_check_discontinuous):
	  Allow very small discontinuities in the timestamps. These we can't
	  do anything useful with anyway (because vorbis's timestamps have
	  only sample granularity), and are commonly produced by elements with
	  minor bugs. Allow up to 1/2 a sample out.
	  Fixes #351742.

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2006-08-24  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
	(play_scrub_toggle_cb), (main):
	Add a checkbox to enable play scrubbing. Makes it possible to disable
	normal scrubbing.

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2006-08-23  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/.cvsignore:
	  make buildbot happy

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
	(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
	(gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
	(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
	(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	(gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
	(gst_ogm_text_parse_strip_trailing_zeroes),
	(gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
	(gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
	  Refactor ogm parse, do better input checking, misc. clean-ups.
	  Cache incoming events and push them once the source pad has
	  been created. Don't pass unterminated strings to sscanf().
	  Strip trailing zeroes from subtitle text output, since they
	  are not valid UTF-8. Don't push vorbiscomment packets on
	  the subtitle text pad. Output perfect streams if possible.

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2006-08-23  Wim Taymans  <wim@fluendo.com>

	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	Waits for tasks to settle down so that we clean up correctly for 
	valgrind.

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
	  Unit test fixes: \377 is more likely to fit into 8 bits than \777;
	  actually return return value in taglists_are_equal.

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Fix crash due to broken bitstream parsing on x86-64: can't make
	  any assumptions about sizeof(struct) due to alignment/packing
	  differences on different architectures. Fixes #351790.

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2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
	(gst_riff_parse_chunk), (gst_riff_parse_file_header),
	(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
	(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
	(gst_riff_parse_info):
	Protect public functions against bad input.
	Do some cleanups.
	Fix documentation.

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Add voxware audio IDs (even if we can't play it) (#351795).

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c:
	(gst_riff_create_video_template_caps),
	(gst_riff_create_audio_template_caps),
	(gst_riff_create_iavs_template_caps):
	  Const-ify some arrays and use G_N_ELEMENTS instead
	  of wasting oodles of RAM on terminator bits.

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_to_vorbiscomment_buffer):
	* tests/check/libs/tag.c: (GST_START_TEST):
	  And the same for _to_vorbiscomment_buffer(): allow
	  id_data_len == 0 for speex.

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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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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.

498 499
2006-08-11  Andy Wingo  <wingo@pobox.com>

500 501
	* gst/tcp/gsttcp.c: Really fix the build?

502 503 504
	* gst/tcp/gsttcp.h: For now, always disable deprecation here --
	fixes the build.

505 506 507 508 509
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.

510 511 512 513 514 515
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).

516 517 518 519 520 521 522 523 524
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).

525 526 527 528 529 530 531 532 533
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).

534 535 536 537 538 539 540 541 542
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).

543 544 545 546 547 548 549 550 551
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.

552 553 554 555 556 557 558 559 560
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.

561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576
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).

577 578 579 580 581
2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

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

582 583 584 585 586 587 588
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).

589 590 591 592 593 594 595 596 597 598
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.

599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614 615
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.

616 617 618 619 620 621 622 623
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).

624 625 626 627 628 629 630
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.

631 632 633 634 635 636 637
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.

638 639 640 641 642 643 644 645 646 647 648 649
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.

650 651 652 653 654 655 656 657 658 659 660
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.

661 662 663 664 665 666 667
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.

668 669 670 671 672
2006-07-26  Thomas Vander Stichele  <thomas at apestaart dot org>

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

673 674 675 676 677 678 679 680
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

681 682 683 684 685 686
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.

687 688 689 690 691 692
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.

693 694 695 696 697 698 699
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.

700 701 702 703 704 705 706 707 708 709 710 711 712 713
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.

714 715 716 717 718 719 720
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.

721 722
2006-07-21  Andy Wingo  <wingo@pobox.com>

723 724 725
	* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
	test to the one in vorbisenc. Also commented out.

726
	* tests/check/pipelines/vorbisenc.c: 
727 728 729
	(test_discontinuity): New test, commented out until Mike lands
	some elite vorbisenc patches.

730 731 732 733 734 735 736
	* 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.

737 738 739 740 741
2006-07-21  Wim Taymans  <wim@fluendo.com>

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

742 743 744 745 746 747 748 749 750 751 752
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.

753 754 755 756 757 758 759
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.

760 761 762 763 764 765
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.

766 767 768 769 770
2006-07-20  Stefan Kost  <ensonic@users.sf.net>

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

771 772 773 774 775 776 777 778 779 780 781 782 783 784
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.

785 786 787 788 789 790 791 792 793
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).

794 795 796 797 798 799 800 801 802
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.

803 804 805 806 807 808 809 810 811
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

812 813 814 815 816 817 818
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.

819 820 821 822 823 824
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).

825 826 827 828 829 830 831 832
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.

833 834 835 836 837 838 839 840 841 842
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.

843 844
2006-07-14  Andy Wingo  <wingo@pobox.com>

845 846 847 848 849 850 851
	* 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.

852 853 854 855 856 857 858 859 860 861 862 863 864 865
	* 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.

866 867 868 869 870
2006-07-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump nano back to CVS

871 872 873 874 875 876 877
=== release 0.10.9 ===

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

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

878 879 880 881 882 883 884
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.

885 886 887 888 889
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

890 891 892 893 894 895 896 897 898
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.

899 900 901 902 903 904 905 906 907 908 909 910 911 912 913 914
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)

915 916 917 918 919 920 921 922
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.

923 924 925 926 927
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.

928 929 930 931 932 933 934 935 936
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.

937 938 939 940 941 942 943 944
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.

945 946 947 948 949 950 951
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.

952 953 954 955 956
2006-07-12  Wim Taymans  <wim@fluendo.com>

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

957 958 959 960 961 962 963
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.

964 965 966 967 968 969 970 971 972 973
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

974 975 976 977 978 979 980
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.

981 982 983 984 985 986 987 988 989 990
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.

991 992 993 994 995 996 997 998 999
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.

1000 1001 1002 1003 1004 1005 1006
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.

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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.

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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).

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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.

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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.

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2006-07-03  Edward Hervey  <edward@fluendo.com>

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

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2006-07-02  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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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

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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.

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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).

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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.

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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.

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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:
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	  Fix --disable-external (can't set conditionals conditionally,
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	  #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.

1514 1515 1516 1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527
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.

1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548
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

1549 1550 1551 1552 1553 1554 1555
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.

1556 1557 1558 1559 1560 1561 1562
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.

1563 1564 1565 1566 1567
2006-06-02  Michael Smith  <msmith@fluendo.com>

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

1568 1569 1570 1571 1572 1573 1574
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.

1575 1576 1577 1578 1579 1580 1581 1582
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.

1583 1584 1585 1586 1587 1588
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

1589 1590 1591 1592 1593 1594 1595 1596 1597 1598 1599 1600 1601 1602
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

1603 1604 1605 1606 1607 1608 1609 1610 1611 1612 1613 1614 1615 1616 1617 1618 1619 1620 1621 1622 1623 1624 1625 1626 1627 1628 1629 1630 1631 1632 1633 1634 1635 1636 1637 1638 1639
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

1640 1641 1642 1643 1644 1645 1646 1647 1648
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.

1649 1650 1651 1652 1653
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

1654 1655 1656 1657 1658 1659 1660
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

1661 1662 1663 1664 1665 1666 1667
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.

1668 1669 1670 1671 1672 1673
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.

1674 1675 1676 1677 1678 1679 1680 1681 1682 1683 1684
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.

1685 1686 1687 1688 1689
2006-05-30  Wim Taymans  <wim@fluendo.com>

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

1690 1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701 1702 1703
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

1704 1705 1706 1707 1708 1709 1710 1711 1712 1713 1714
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.

1715 1716 1717 1718 1719 1720 1721 1722
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.

1723 1724 1725 1726 1727
2006-05-29  Wim Taymans  <wim@fluendo.com>

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

1728 1729 1730 1731 1732 1733 1734 1735 1736 1737 1738 1739
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.

1740 1741 1742 1743 1744 1745 1746 1747 1748 1749 1750 1751 1752 1753 1754 1755 1756 1757 1758
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).

1759 1760 1761 1762 1763 1764 1765 1766 1767
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.

1768 1769 1770 1771 1772
2006-05-28  Stefan Kost  <ensonic@users.sf.net>