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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (no_more_pads):
	Fix compilation.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (new_caps):
	Remove g_print

	* gst/playback/gstplaybin.c:
	Add some docs.

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2006-09-29  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Re-enable cddabasesrc test to see if it works again
	  now.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (setup_subtitle),
	(gen_source_element):
	Handle invalid URIs a bit more gracefully.

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2006-09-29  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c:
	  Remove obsolete comment.

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2006-09-29  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
	(gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
	(gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
	(gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
	(gst_ogg_mux_collected):
	  Commit patch from James "Doc" Livingston, adds proper EOS handling
	  in oggmux. GStreamer can, for the first time ever, create a valid
	  Ogg file! Yay!

	* tests/check/pipelines/oggmux.c: (check_chain_final_state),
	(oggmux_suite):
	  Reenable tests now that they pass.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
	Stop reading commands when EOF (we read 0) as well.

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2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
	(close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
	(find_dynamic), (unlinked), (close_link):
	Implement delayed caps linking needed for element with a lot of
	different caps on the src pads that get fixed at runtime.
	Improve management of dynamic elements.

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	(group_destroy), (group_commit), (check_queue), (queue_overrun),
	(gen_preroll_element), (remove_groups), (unknown_type),
	(add_element_stream), (no_more_pads_full), (no_more_pads),
	(sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
	(new_decoded_pad), (setup_subtitle), (array_has_value),
	(gen_source_element), (source_new_pad), (has_all_raw_caps),
	(analyse_source), (remove_decoders), (make_decoder),
	(remove_source), (setup_source), (finish_source), (prepare_output),
	(gst_play_base_bin_change_state):
	* gst/playback/gstplaybasebin.h:
	Use more _CAST instead of full type checking casts.
	Small cleanups, plug some leaks.
	Handle dynamic sources.
	Add some helper functions to create lists of strings used for
	blacklisting and other stuff.
	Refactor some code dealing with analysing the source.
	Re-enable sources without pads (like cd:// or other selfcontained
	elements).

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2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	When we have a timestamp, we can still perform clipping.
	When we have no clock, we must play the sample ASAP.

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2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	Set caps on outgoing buffers.

	* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
	(gst_video_rate_event), (gst_video_rate_chain):
	* gst/videorate/gstvideorate.h:
	Fix videorate some more. Fixes #357977

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2006-09-28  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/adder.c: (adder_suite):
	  Don't set timeout to 6 seconds when we're running
	  in valgrind ... (and how is 6 seconds longer than
	  the default anyway?)

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2006-09-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	(gst_audio_rate_sink_event), (gst_audio_rate_convert),
	(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
	Keep sink and src segment to keep track of time and support more
	input formats.
	Fix bogus next_offset and run_time calculation, don't understand how
	this could have worked before. Fixes #357976.
	Remove some unneeded vars.

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2006-09-28  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (remove_sinks):
	  Only remove visualisation from visbin if there is a visbin (or:
	  don't throw warnings when closing totem without playing a file).

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2006-09-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Add some more info in a WARNING.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	Handle PAUSE in create function, use new -core addition to
	wait for playing. Fixes pausing and resuming capture from an
	audiosrc.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	Constify some more.
	Caller supports interrupted reads now.

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2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Another attempt to make the gen64 buildbot happy.

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2006-09-27  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>

	* ext/libvisual/visual.c: (gst_visual_clear_actors),
	(gst_visual_chain), (gst_visual_change_state):
	  Libvisual plugin was not passing audio data to libvisual 0.4.0 
	  correctly. Fixes #357800

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2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
	  Add timeout to _get_state() so we see which pipeline it is
	  that causes trouble on the gen64 build bot.

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2006-09-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
	(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
	(gst_base_rtp_depayload_set_gst_timestamp):
	the source pad always uses fixed caps.

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Wim Taymans committed
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2006-09-27  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudioclock.c:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	Added docs for the audio libs.

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2006-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Temporarily disable test that fails on the bots for unknown reasons.

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2006-09-26 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	Moved AudioCodecType into priv
	Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes

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2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
	(add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
	(is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
	(new_pad):
	Cleanups and small leak fixes.
	Added Depayloaders to valid list of autopluggable elements.

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2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_vis_blocked), (gst_play_bin_set_property),
	(gen_video_element), (gen_text_element), (gen_audio_element),
	(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
	(gst_play_bin_set_clock_func), (gst_play_bin_change_state):
	Detect NO_PREROLL state change returns and disable clock distribution to
	the sinks so that sync is disabled.
	Avoid some type checking and do simple casts instead.
	Small cleanups, fix some FIXMEs.
	Be more robust when linking user specified elements, catch an report
	errors. Fixes #357404.
	Fix some leaks in the error paths.

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2006-09-25  Stefan Kost  <ensonic@users.sf.net>

	* ChangeLog:
	  ChangeLog surgery for missing bug-number

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/playback/test.c:
	  Fix compilation with uClibc and -Werror (#357591).

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse dates that are followed by a time as well (#357532).

	* tests/check/libs/tag.c: (test_vorbis_tags):
	  Add unit test for this.

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2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(gst_audio_convert_transform_caps):
	* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
	* gst/videotestsrc/videotestsrc.h:
	  A few array const-ifications.

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2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  See if this makes the build bots happy.

	* tests/check/libs/cddabasesrc.c:
	  UTF8-ise my name.

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2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/subparse/samiparse.c: (handle_start_font),
	(fix_invalid_entities):
	  More case-insensitivity for certain tags; recognise entities with
	  decimal codes as special entities as well (#357330).

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2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/Makefile.am:
	  Need to build tag directory before cdda.

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2006-09-23  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/cdda/Makefile.am:
	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_base_init):
	* gst-libs/gst/cdda/gstcddabasesrc.h:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
	(gst_tag_register_musicbrainz_tags):
	  Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
	  depend on libgsttag. This is required so we can extract/read tags like
	  DISCID without depending on libgstcddabasesrc (which used to register
	  them).

	* gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
	  tags (also see #347848).

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
	  Log vorbis comments we are actually writing. Const-ify array.

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (gen_preroll_element):
	Improve buffering a bit by avoiding a deadlock because we cannot assume
	the underrun is always called.

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	Patch by: Young-Ho Cha <ganadist at chollian dot 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):
	Added MPEG-4 AAC and id and caps. Fixes #357289
	Added WMA9 Lossless id.

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2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c:
	  Fix misleading docs addition.

	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Get rid of compiler warning the right way.

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_finalize),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_push_full),
	(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
	(gst_base_rtp_depayload_process),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_queue_release):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Small cleanups.
	Fix some leaks.
	Refactored the process method and added methods to push from the process
	vmethod.
	Use _scale functions.
	API: gst_base_rtp_depayload_push_ts
	API: gst_base_rtp_depayload_push

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	timestamps are uint.

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2006-09-22  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/xoverlay.c:
	  Remove unused statement from doc example.

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2006-09-21  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/videoorientation.c:
	(gst_video_orientation_iface_init),
	(gst_video_orientation_get_hflip),
	(gst_video_orientation_get_vflip),
	(gst_video_orientation_get_hcenter),
	(gst_video_orientation_get_vcenter),
	(gst_video_orientation_set_hflip),
	(gst_video_orientation_set_vflip),
	(gst_video_orientation_set_hcenter),
	(gst_video_orientation_set_vcenter):
	  Add since tags to new API docs, ChangeLog surgery (forgot API keyword
	  in ChangeLog)

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2006-09-21  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
	(create_rgb_conversions), (rgb_conversion_free),
	(right_shift_colour), (fix_expected_colour), (check_rgb_buf),
	(got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
	  Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
	  but disable for now since it doesn't pass (something wrong with
	  RGBA somewhere).

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (group_commit),
	(queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
	(queue_out_of_data), (gen_preroll_element),
	(preroll_remove_overrun), (probe_triggered):
	Refactor handling of overrun detection.
	Separate handling of group completion and deadlock detection when doing
	network buffering. This should fix some deadlocks that were not detected
	because the group was completed.
	Add more comments, improve debugging.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/gdpdepay.c: (GST_START_TEST):
	* tests/check/libs/audio.c:
	Some more compilation fixes.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	Early morning compilation fix.

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Wim Taymans committed
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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/gdpdepay.c: (GST_START_TEST):
	* tests/check/elements/multifdsink.c: (GST_START_TEST):
	* tests/check/elements/videorate.c: (GST_START_TEST):
	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (eos_buffer_probe):
	Fix some warnings.

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2006-09-20  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
	  Handcrafted merge to help CVS understanding what I changed and what
	  not.

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2006-09-20  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_get_times):
	  change colorkey behaviour back according to #354773 comment 6/7

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2006-09-19  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
	(gst_multi_fd_sink_recover_client),
	(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
	(gst_multi_fd_sink_get_property):
	* gst/tcp/gstmultifdsink.h:
	  Implement stubbed out properties unit-type, units-soft-max,
	  units-max, to allow specifying maximum sizes in units other than
	  buffers.
	  Fixes #355935

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Reorder the audio formats a bit for clarity.
	Detect and create caps for MSGSM and MSN (WAV49).
	Fixes #356596.

	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
	Small cleanups, move error handling out of normal flow for clarity.

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2006-09-18  Stefan Kost  <ensonic@users.sf.net>
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	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/videoorientation.c:
	(gst_video_orientation_get_type),
	(gst_video_orientation_iface_init),
	(gst_video_orientation_get_hflip),
	(gst_video_orientation_get_vflip),
	(gst_video_orientation_get_hcenter),
	(gst_video_orientation_get_vcenter),
	(gst_video_orientation_set_hflip),
	(gst_video_orientation_set_vflip),
	(gst_video_orientation_set_hcenter),
	(gst_video_orientation_set_vcenter):
	* gst-libs/gst/interfaces/videoorientation.h:
460
	  API: Add new interface to control video orientation (fixes #354908)
461

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2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* gst/videotestsrc/gstvideotestsrc.c:
	  Use G_UNLIKELY in _create and log one more detail.
	  
	(gst_video_test_src_get_times), (gst_video_test_src_create):
	* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
	  Use gst_util_uint64_scale_int in _get_times().

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2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
	  Give better warning message (add object and detail).

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2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_get_times):
	  xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
	  #354773), use gst_util_uint64_scale_int in _get_times()

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2006-09-18  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
	  Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
	  always true, leading to dropping all timestamps.

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2006-09-18  Stefan Kost  <ensonic@users.sf.net>

	* ext/libvisual/visual.c: (gst_vis_src_negotiate),
	(gst_visual_chain), (gst_visual_change_state):
493
	  update to work also with libvisual 0.4 API, fix double unref (#355914)
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	* tools/gst-launch-ext.1.in:
	* tools/gst-visualise.1.in:
	  remove references to old man-pages

	* tests/examples/seek/seek.c: (main):
	  add real meadi-buttons, add tool-tips for the seek-options, arrange
	  seek options in a table

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2006-09-18  Michael Smith  <msmith@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
	(gst_ogg_mux_push_buffer):
	  Don't generate out-of-order timestamps from oggmux, instead clamp
	  output timestamps to be >= the previously output ts.
	  Fixes #355595

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2006-09-18  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_multi_fd_sink_class_init):
	  Updates, fixes, and typo corrections for multifdsink. No functional
	  changes.

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2006-09-17  Michael Smith  <msmith@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
	  Don't crash on truncated files - check that we got an 8 byte buffer
	  before trying to memcmp it.

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2006-09-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (get_active_source):
	  Make stream-switching appear instant to the application
	  (ie. make sure that a g_object_get on 'current-foo' returns
	  the stream previously set with g_object_set(). Totem needs
	  this to update stream-related meta-info (like audio-codec)
	  correctly when switching streams.

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2006-09-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
	(gst_alsa_mixer_ensure_track_list):
	  Try harder to guess which mixer track is the master mixer
	  track (instead of just taking the first one that has a pvolume).
	  Fixes #342228.

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2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	(gst_audio_convert_transform_caps):
          Get structure-name just once.

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2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audioresample.c: (GST_START_TEST):
	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	* tests/check/elements/volume.c: (GST_START_TEST):
	* tests/check/elements/vorbisdec.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
	(test_pipeline), (GST_START_TEST):
	* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
          Fix big batch of compiler warnings.

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2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* ext/gnomevfs/gstgnomevfssrc.c:
          Add docs about icydemux usage in connection with gnomevfssrc

	* ext/libvisual/visual.c:
	* ext/ogg/gstoggaviparse.c:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggparse.c:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst/audiorate/gstaudiorate.c:
	  More G_OBJECT macro fixing.

	* gst/audiotestsrc/gstaudiotestsrc.h:
          Fix wrong info in header due to copy & paste

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
	(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
	(gst_base_audio_src_create), (gst_base_audio_src_change_state):
	Do the delay calculation in the source/sink base classes as this is
	specific for the capture/playback mode.
	Try to fixate a bit better, like round depth up to a multiple of 8
	bigger than width.
	Handle underruns correctly by marking DISCONT on buffers and adjusting
	timestamps to handle the gap.
	Set offset/offset_end correctly on buffers.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
	(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	Remove resync and underrun recovery from the ringbuffer.
	Fix ringbuffer read code on under/overrun.

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (fill_buffer), (check_queue),
	(queue_threshold_reached), (gst_play_base_bin_set_property),
	(gst_play_base_bin_get_property):
	* gst/playback/gstplaybasebin.h:
	Don't use a 0 low watermark when buffering, it is catching starvation
	way too late. Instead, use a 3 second queue with 30 and 95
	percent low/high watermarks. 
	Added queue-min-threshold property to configure low watermark.
	Use new _buffering message API.
	Make queue_threshold variable big enough to store a uint64 time value.
	API: playbin::queue-min-threshold property.

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	We require 0.10.10.1 now because of _wait_preroll().

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Use gst_base_sink_wait_preroll().

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
	* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
	Use DEBUG_OBJECT more.

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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=== release 0.10.10 ===

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2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Michael Smith <msmith at fluendo dot com>

	* gst/tcp/gstmultifdsink.c: (is_sync_frame),
	(gst_multi_fd_sink_client_queue_buffer),
	(gst_multi_fd_sink_new_client):
	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(multifdsink_suite):
	  Fix implementation of sync-method 'next-keyframe'
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	  Closes #354594
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2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Wim Taymans <wim at fluendo dot com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	This patch removes the RANDOM flag that was incorrectly introduced with
	revision 1.91.  Fixes #354590

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2006-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Random variation in Makefile line to see if it makes the
	  gen64-base-full bot any happier.

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c: (oggmux_suite):
	  Disable test that fails at the moment (killed after timeout).

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: James Livingston  <doclivingston at gmail.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/.cvsignore:
	* tests/check/pipelines/oggmux.c: (get_page_codec),
	(check_chain_final_state), (fail_if_audio), (validate_ogg_page),
	(eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
	(test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
	(test_theora_vorbis), (oggmux_suite):
	  Add simple unit test for oggmux from #337026 with checking for the
	  EOS flags disabled for the time being.

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2006-09-04  Wim Taymans  <wim@fluendo.com>

	patch by: Alessandro Dessina <alessandro nnva org>

	* ext/ogg/gstoggmux.c:
	Add cmml caps to oggmux. Fixes #353912

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2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Returning a return value often helps. In this case, we
	  don't need the return value anyway, so just get rid of it.
	  Should make build bots much happier.

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2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
	(paint_get_structure), (gst_video_test_src_get_size),
	(gst_video_test_src_smpte), (gst_video_test_src_snow),
	(gst_video_test_src_unicolor), (paint_setup_AYUV),
	(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
	(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
	* gst/videotestsrc/videotestsrc.h:
	  Add support for AYUV and the various RGBA formats. Initialise
	  fields of paintinfo structs allocated on the stack.

	* tests/check/elements/videotestsrc.c: (right_shift_colour),
	(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
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	(check_rgb_buf), (videotestsrc_suite):
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	  Add unit tests for videotestsrc's RGB output.

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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
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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	  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
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	  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.
841

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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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Wim Taymans committed
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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):
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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	  API: GstPlayBin::connection-speed
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	  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)
Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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	(theora_parse_push_buffer, theora_parse_change_state):
	API: GstTheoraParse::synchronization-points
	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_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:
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	  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.

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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).

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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.

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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):
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	  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
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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.

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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).

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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.

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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.

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

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	* 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.

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	* 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>

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

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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"

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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.

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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

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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.

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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.

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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.

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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.

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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.

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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.