ChangeLog 1.17 MB
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2007-06-29  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
	Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
	format, as produced by some dc1394 cameras like the iSight.
	See http://www.fourcc.org/yuv.php#IYU1

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2007-06-28  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	(check_default), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_dithering_get_type),
	(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
	(gst_audio_convert_init), (gst_audio_convert_set_caps),
	(gst_audio_convert_set_property), (gst_audio_convert_get_property):
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	(gst_audio_quantize_setup_noise_shaping),
	(gst_audio_quantize_free_noise_shaping),
	(gst_audio_quantize_setup_dither),
	(gst_audio_quantize_free_dither),
	(gst_audio_quantize_setup_quantize_func),
	(gst_audio_quantize_setup), (gst_audio_quantize_free):
	* gst/audioconvert/gstaudioquantize.h:
	Implement dithering and noise shaping in audioconvert. By default now
	TPDF dithering (and no noise shaping) will be used when converting
	from a higher bit depth to 20 bit depth or smaller, otherwise
	everything will be as it is now.
	For the last audioconvert in a pipeline it would make sense to
	use some kind of noise shaping, enabling it by default for all
	conversions would give undesired results though. Fixes #360246.
	* tests/check/elements/audioconvert.c: (setup_audioconvert),
	(GST_START_TEST):
	Adjust unit test for the new audioconvert.

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2007-06-28  Wim Taymans  <wim@fluendo.com>

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

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2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
	Small debug improvement.

	* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
	(plugin_init):
	Tweak the rate estimation period.
	When calculating the buffer filledness in rate estimation mode, don't
	mix it with other metrics.

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2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
	(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
	When creating the groups, allow for a 5 second, unlimited buffers
	preroll phase after which we expose the group.
	When the group is exposed, use a small number of buffers up to a 2
	second limit. Also disconnect the overrun signal from multiqueue when we
	exposed the group because it is not needed anymore.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
	  to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
	  (#451707); also, output some debugging info when dealing with
	  freeform strings.

	* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
	  Add unit test for the above.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
	  Add description for Windows Media RTP caps.

	* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
	  Remove RTP fields that don't define the format from caps.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	  Skip empty buffers, but not empty header buffers. That way the original
	  vorbisdec unit test still passes (#451145); also, take into account
	  that those empty packets might carry a granulepos.

	* tests/check/Makefile.am:
	* tests/check/elements/vorbisdec.c:
	(_create_codebook_header_buffer), (_create_audio_buffer),
	(GST_START_TEST), (vorbisdec_suite):
	  Add unit test that sends an empty packet.

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2007-06-27  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	Don't error out on 0-sized packets, just emit a warning because this is
	not a fatal error. Fixes #451145.

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2007-06-25  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	  Update docs with caps info.

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

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

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2007-06-23  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
	The chain should be freed if we error out here, else it will leak.
	* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
	(cleanup_decodebin):
	Don't forget to *properly* remove the signals, else it will leak.

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

	* MAINTAINERS:
	Updating all the maintainers files

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2007-06-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
	(main):
	  Destroy and recreate parse-launch based pipeline after stop to be able
	  to play again. Reorder some code and add more comments.

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2007-06-20  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin2.c: (analyze_new_pad):
	When handling a delayed-caps notification case, mark
	the group as dynamic so that the nbdynamic count is
	incremented and decremented correctly. Fixes: #449156
	Patch by: Wim Taymans <wim@fluendo.com>

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

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

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2007-06-19  Michael Smith <msmith@fluendo.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Change minimum rate back to 1000 to allow low-sample-rate wav files
	  to play back.

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

	* po/vi.po:
	  Update translations.

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

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

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

	* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
	  Update tmpbuf for all neccesary rows, not just one, as is required
	  when downscaling.
	  Fixes #402076.

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

	* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
	(eos_buffer_probe):
	  Add a test that ensures we set DELTA_UNIT on all non-header,
	  non-video buffers, if we have a video stream.
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
	(gst_ogg_mux_process_best_pad):
	  Move setting delta_pad to earlier, where we inspect all pads, so
	  that leading audio pages don't get DELTA_UNIT unset if they come
	  before the first DELTA_UNIT from video pages. Fixes the newly-added
	  test. Fixes #385527.

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

	* tests/check/pipelines/streamheader.c: (streamheader_suite):
	  Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
	  fails on the p5-ppc64 build bot and the failure looks like it is due
	  to the same issue as #348114, ie. a compiler bug.

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

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

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

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
	Fix compilation on mingw. Fixes #446972.

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

	Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (update_buffering),
	(gst_queue_locked_enqueue):
	Fix a division by zero when the max percent is <= 0. Fixes #446572.
	also update the buffering status when receiving events. Fixes #446551.

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

	Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_peer_query),
	(gst_queue_handle_src_query):
	Wait for preroll before attempting to forward a duration query upstream.
	Fixes #445505.

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

	* gst-libs/gst/rtp/gstbasertpdepayload.c: 
	(gst_base_rtp_depayload_set_gst_timestamp):
	Use G_GINT64_CONSTANT macro for int64 constant.
	* win32/common/libgstinterfaces.def:
	* win32/common/libgsttag.def:
	Add new exported functions.

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

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
	  The BOS page of the first Dirac video stream needs to come before
	  the BOS page of any Vorbis streams or other audio streams, just like
	  it is with Theora.

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

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

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

	Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_init),
	(gst_queue_handle_sink_event), (gst_queue_chain),
	(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
	(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
	(gst_queue_src_activate_pull):
	Add pull based scheduling and fix some deadlocks. Fixes #444523.
	Does not yet completely work because duration queries upstream won't
	block yet.

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

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

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

	* configure.ac:
	check for large file support.

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

	Based on a patch by Sven Arvidsson <sa at whiz dot se>:

	* gst/subparse/gstsubparse.c: (parse_subrip),
	(subviewer_unescape_newlines), (parse_subviewer),
	(gst_sub_parse_data_format_autodetect),
	(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
	* gst/subparse/gstsubparse.h:
	Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	Add a unit test for both SubViewer formats.

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

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

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

	* gst/playback/gstqueue2.c: (gst_queue_have_data),
	(gst_queue_create_read), (gst_queue_read_item_from_file),
	(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
	Include stdio to define fseeko.

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

	Patch by: Edward Hervey  <edward@fluendo.com>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
	(gst_v4lsrc_query):
	Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.

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

	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  our own implementation.

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	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state):
	Handle timestamp wraparound.

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

	* gst/playback/gsturidecodebin.c: (no_more_pads_full),
	(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
	(gst_uri_decode_bin_change_state):
	Make sure we name srcpads uniquely even when using different internal
	decodebins.
	Signal no-more-pads when no more dynamic elements exist.
	Remove pads on cleanup.

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

	Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_class_init),
	(gst_queue_init), (gst_queue_finalize),
	(gst_queue_write_buffer_to_file), (gst_queue_have_data),
	(gst_queue_create_read), (gst_queue_read_item_from_file),
	(gst_queue_open_temp_location_file),
	(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_is_empty), (gst_queue_is_filled),
	(gst_queue_change_state), (gst_queue_set_temp_location),
	(gst_queue_set_property):
	Add support for filebased buffering. Fixes #441264.

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

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
	(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
	(caps_notify_group_cb), (gst_decode_group_new),
	(gst_decode_group_free):
	Add support for delayed caps fixation when autoplugging.
	Optimize cases where a multiqueue is not needed/wanted, like right after
	anything that is not a demuxer.

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

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
	consideratly speedup ogg chain detection by not trying to find a base
	timestamp for skeleton streams. 

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

	* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
	(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_remove_flush),
	(gst_multi_fd_sink_remove_client_link),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_handle_clients):
	* gst/tcp/gstmultifdsink.h:
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	Add support for remove_flush.
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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* docs/design/draft-keyframe-force.txt:
	* ext/theora/theoraenc.c: (theora_enc_sink_event),
	(theora_enc_chain):
	Add draft design for forcing keyframes in encoders and implement in
	theoraenc.

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

	* configure.ac:
	  Back to CVS

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=== release 0.10.13 ===

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

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

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	In riff, the depth is stored in the size field but it just means that
	the least significant bits are cleared. We can therefore just play
	the sample as if it had a depth == width. Fixes: #440997

	Patch by: Wim Taymans <wim@fluendo.com> 
	Patch by: Sebastian Dröge  <slomo@circular-chaos.org>

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	* gst-libs/gst/floatcast/floatcast.h:
	Define inline when needed on win32 builds. Fixes: #441295
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	Patch by: Sebastien Moutte  <sebastien@moutte.net>
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2007-05-29  Wim Taymans  <wim@fluendo.com>

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

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

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	(gst_alsa_mixer_free), (gst_alsa_mixer_update),
	(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
	(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
	(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_interface_supported),
	(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
	(gst_alsa_mixer_element_set_property),
	(gst_alsa_mixer_element_get_property),
	(gst_alsa_mixer_element_change_state):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
	* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
	(gst_mixer_option_changed):
	* gst-libs/gst/interfaces/mixer.h:
	Revert commits towards #152864 made so far. We'll pick it up again
	after the 0.10.13 release.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	After an interrupt (PAUSED/flush) assume that the next sample should not
	be aligned to the previous sample. Fixes #417992.

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Don't add channels and rate fields to the template caps for
	  audio/x-dts, as wavparse might not always be able to set them,
	  which would then lead to 'caps are not a real subset of the
	  template caps' warnings.

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

	* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
	Handle unknown or invalid pads without crashing, as might occur if
	a media file like an mp3 is specified as a subtitle file.
	Fixes: #410039

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

	* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
	(setup_sinks):
	Block the subtitle bin output queue before ghosting it and linking,
	then unblock after. This avoids spurious not-linked errors caused 
	by the queue starting up (because it gets linked when it is ghosted). 
	Fixes: #350299

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

	* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
	Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
	file. Avoids flukes where the input gets typefound to some valid but
	useless type.

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

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
	(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
	  Add unit test for gnomevfssink seeking and position reporting for
	  file:// URIs.

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

	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
	(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
	(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
	* ext/gnomevfs/gstgnomevfssink.h:
	  Fix position reporting, especially after a seek (from upstream),
	  see #412648.

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

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

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	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Specify the full valid range for MP3 samplerates. Fixes a regression
	caused by extra header checks since the last release.

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

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	Fix a locking-order bug I introduced with my changes the other day.
	Patch by Mike Smith.

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

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

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

	* ext/cdparanoia/gstcdparanoiasrc.c:
	(gst_cd_paranoia_src_read_sector):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	Small cleanups.

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

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_set_gst_timestamp):
	Add some FIXME

	* gst/playback/gstdecodebin.c: (queue_underrun_cb):
	And some debug info when a FIXME path is hit.

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

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init),
	(gst_base_rtp_audio_payload_init),
	(gst_base_rtp_audio_payload_finalize),
	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
	(gst_base_rtp_payload_audio_handle_event):
	Some cleanups, remove minptime property as it is now in the parent
	class.
	Override parent class event function.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_class_init), (gst_basertppayload_init),
	(gst_basertppayload_event), (gst_basertppayload_set_property),
	(gst_basertppayload_get_property):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Add min-ptime property.
	Add handle-event vmethod. Fixes #415001.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_change_state):
	  Fix typo in comment.

	* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
	  free_dynamics, pad_probe, close_pad_link, try_to_link_1,
	  get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
	  close_link):
	* gst/playback/gstplaybin.c (gst_play_bin_set_property,
	  gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
	  Remove trailing whitespaces in comments.

	* gst/volume/Makefile.am:
	  Fix tabs.

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	patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
	  set_option, get_option, _gst_reserved):
	  Revert reordering functions (keep ABI).

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

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
	(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
	(gst_ximagesink_show_frame):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_show_frame):
	When we create our own window, indicate that we handle the 
	WM_DELETE client message from the window manager, so that it won't 
	kill our window (and our app) along with it. Handle ClientMessage,
	post an error on the bus, and close the window. Further buffers
	arriving will result in a FlowError because the window has been
	destroyed.

	Fixes: #393975

	Clean up the X event handling loop and make them the same for
	both xvimagesink and ximagesink while I'm at it.

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2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
	Make decodebin2 autoplug depayloaders too.

	* gst/playback/gsturidecodebin.c: (source_new_pad):
	Set the newly created decoder in a usable state when autoplugging a
	dynamic source such as RTSP.

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2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c: (cb_probe):
	  Ignore video-codec tag for audio streams and ignore audio-codec tags
	  for video streams. Should make codec name collection a bit more
	  robust against sloppy demuxers that send tag events containing both
	  tags down each pad.

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2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (update_rates):
	Tweak the buffering thresholds a little.
	Update the buffer size with the previously calculate rate instead of
	only when we calculate a new rate so that we get smoother buffering
	updates.

	* gst/playback/Makefile.am:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
	(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
	(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
	(gst_uri_decode_bin_get_property), (unknown_type),
	(add_element_stream), (no_more_pads_full), (no_more_pads),
	(source_no_more_pads), (new_decoded_pad), (array_has_value),
	(gen_source_element), (has_all_raw_caps), (analyse_source),
	(remove_decoders), (make_decoder), (remove_source),
	(source_new_pad), (setup_source), (decoder_query_init),
	(decoder_query_duration_fold), (decoder_query_duration_done),
	(decoder_query_position_fold), (decoder_query_position_done),
	(decoder_query_latency_fold), (decoder_query_latency_done),
	(decoder_query_seeking_fold), (decoder_query_seeking_done),
	(decoder_query_generic_fold), (gst_uri_decode_bin_query),
	(gst_uri_decode_bin_change_state), (plugin_init):
	New element that intergrates a source, optional buffering element and
	decodebin.

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2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Bump libtheora requirement to 1.0alpha5 for the pixformat check
	  (also has a .pc file, so we don't need the fallback check any
	  longer). Fixes #438840.

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2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
	(apply_segment), (apply_buffer), (update_buffering),
	(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_filled),
	(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
	(plugin_init):
	fix build.

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2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/playback/Makefile.am:
	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
	(gst_queue_getcaps), (gst_queue_bufferalloc),
	(gst_queue_acceptcaps), (update_time_level), (apply_segment),
	(apply_buffer), (update_buffering), (reset_rate_timer),
	(update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_empty),
	(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
	(gst_queue_loop), (gst_queue_handle_src_event),
	(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
	(gst_queue_src_activate_push), (gst_queue_change_state),
	(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
	On our way to playbin2 this is the new network queue that does buffering
	all by itself using high and low watermarks. It can also measure up and
	downstream bandwidth to optimally size the queue.

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2007-05-17  Michael Smith <msmith@fluendo.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
	  Use the segment->last_stop value to calculate the next timestamp to
	  generate after a seek; not the segment->start value.

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2007-05-15  David Schleef  <ds@schleef.org>

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

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2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
	Some more chained streaming ogg timestamp fixes.

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2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_handle_page):
	Add some FIXMEs.
	Fix chain start/stop segment handling based on patch by
	<ahalda at cs dot mcgill dot ca> see #320984.

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2007-05-15  Michael Smith <msmith@fluendo.com>

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

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2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track):
	  Apply some of the cleanup Tim suggested in #152864 afterwards.

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2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
	  _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
	  gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
	  gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
	  gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
	  gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
	* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
	* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
	  gst_alsa_mixer_element_interface_supported,
	  gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
	  gst_alsa_mixer_element_set_property,
	  gst_alsa_mixer_element_get_property,
	  gst_alsa_mixer_element_change_state):
	* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
	* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
	  gst_mixer_option_changed):
	* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
	  volume_changed, option_changed, _gst_reserved):
	  Implement notification for alsamixer. Fixes #152864

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2007-05-14  David Schleef  <ds@schleef.org>

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

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2007-05-12  David Schleef  <ds@schleef.org>

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

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	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Parse and use additional caps fields as described in updated
	application/x-rtp caps spec.

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2007-05-12  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_collect_chain_info):
	If there is a stream in a chain without any data packets, ignore the
	stream in the total length calculations. Might be related to #436820.

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2007-05-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
	(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
	(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
	(mpeg_video_type_find), (mpeg_video_stream_type_find),
	(plugin_init):

	Consolidate and re-work our mpeg system stream detection to probe
	more packets and produce a higher confidence result. Fixes a
	regression caused by lowering the typefind probability last year
	- related to bug #397810. Remove the redundant MPEG-1 specific 
	typefind function, as the new one detects both MPEG-1 & MPEG-2
	happily.

	Also cleanup the MPEG elementary and MPEG-TS detection functions a
	little. 

	Tested against my media test directory, with some improvements and
	no regressions.

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2007-05-10  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
	(queue_out_of_data):
	Connect to the new queue "pushing" signal instead of the broken
	"running" one.

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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	Move variable declaration before the first instruction.
	* gst/videotestsrc/videotestsrc.c:
	Define M_PI if it's not defined yet.
	* win32/common/libgstrtp.def:
	Add new exported functions.

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2007-05-09  Michael Smith <msmith@fluendo.com>

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

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

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

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2007-05-08  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
	  gst_adder_change_state):
	* gst/adder/gstadder.h (bps, offset, collect_event, segment,
	  segment_pending, segment_position, segment_rate):
	  Handle playback-rate on adder.

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2007-05-07  Michael Smith <msmith@fluendo.com>

	* ext/theora/gsttheoradec.h:
	* ext/theora/theoradec.c: (gst_theora_dec_reset),
	(theora_dec_sink_event), (theora_handle_comment_packet),
	(theora_handle_type_packet), (theora_dec_change_state):
	  Don't push events (newsegment, tags) before initialising the
	  decoder.
	  This is neccesary for seeking to work correctly in gnonlin.

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2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst/adder/gstadder.c:
	* gst/audiotestsrc/gstaudiotestsrc.c
	  (gst_audio_test_src_create_white_noise):
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
	  VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
	  volume_sink_template, volume_src_template, gst_volume_init,
	  volume_process_double, volume_process_int16,
	  volume_process_int16_clamp):
	  Doc fixes and formatting.

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2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  Minimal check for volume's GstController usability; also another
	  test for #422295.

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2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix it so that it (a) makes sense and (b) doesn't break
	  everything cdda-related including the unit test.

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2007-05-04  Stefan Kost  <ensonic@users.sf.net>

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

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2007-05-03  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
	  When XShm is not available, we might get row strides that are not
	  rounded up to multiples of four; this is bad, because virtually
	  every RGB-processing element in GStreamer assumes rowstrides are
	  rounded up to multiples of four, so let's allocate at least enough
	  memory to avoid crashes in this case. The image will still be
	  displayed distorted though if this happens, so that still needs
	  fixing (maybe by allocating a bigger image with an 'even' width
	  and then clipping it appropriately when rendering - something for
	  Xlib aficionados in any case).

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2007-05-03  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If a buffer doesn't have a timestamp, assume it's contiguous with
	  the previous buffer, and synthesise timestamps appropriately.

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2007-05-03  Edward Hervey  <edward@fluendo.com>

	* tests/check/elements/videorate.c: (GST_START_TEST):
	Set buffer timestamp to a valid value in order to test the buffer
	really does stay in videorate.

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2007-05-03  Edward Hervey  <edward@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	There is no sensible way to handle incoming buffers which don't have a
	valid timestamp. We therefore discard them and wait for the next one.

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2007-05-01  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-29  Wim Taymans  <wim@fluendo.com>

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

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2007-04-27  Julien MOUTTE  <julien@moutte.net>

	* ext/theora/theoradec.c: (_theora_granule_time),
	(theora_dec_push_forward), (theora_handle_data_packet),
	(theora_dec_decode_buffer): Calculate buffer duration correctly
	to generate a perfect stream (#433888).
	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont): Glib provides ABS.

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2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix RB block parsing and writing.
	Add support for constructing BYE packets.

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2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
	(gst_base_audio_src_create):
	* po/POTFILES.in:
	  When posting a warning message because samples were dropped, post
	  something more intelligible than he default error message for clock
	  errors which is just confusing in this context (#432984).

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2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
	(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
	(read_packet_header), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
	(gst_rtcp_packet_sdes_get_item_count),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_entry),
	(gst_rtcp_packet_sdes_next_entry),
	(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Implement code to write SR, RR and SDES packets.

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

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

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

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2007-04-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
	Initalize the AudioConvertCtx with zeroes, otherwise it will contain
	pointers to random memory which are passed to g_free() when
	audio_convert_prepare_context() is called the first time.

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams <dcbw redhat com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  Don't leak incoming buffer if gst_pad_push() returns a
	  non-OK flow. Fixes #432755.
	 
	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	  Unit test for the above by Yours Truly.

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2007-04-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
	(gst_adder_sink_event), (gst_adder_collected):
	  Fix non-flushing segmented seeks, Fixes #340060 for me

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Olivier Crete  <tester at tester ca>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init),
	(gst_base_rtp_audio_payload_init),
	(gst_base_rtp_audio_payload_dispose):
	  Chain up to parent class in dispose function; get rid of
	  unnecessary 'diposed' flag in private structure (#415001).

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init):
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	  Some minor docs fixes and additions; also add missing 'Since' bits.

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Zeeshan Ali  <zeenix gmail com>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
	(gst_base_rtp_audio_payload_push):
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  The recently-added gst_base_rtp_audio_payload_push() should take an
	  object of type GstBaseRTPAudioPayload as first argument (#431672).

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/subparse/gstsubparse.c:
	* gst/subparse/samiparse.c:
	  Use GST_DISABLE_XML here

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_navigation_send_event):
	* sys/xvimage/xvimagesink.h:
	  Include stdlib.h when using atoi.
	  
	* tests/check/elements/playbin.c: (playbin_suite):
	  Use GST_DISABLE_REGISTRY here

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2007-04-19  Michael Smith  <msmith@fluendo.com>

	* ext/theora/gsttheoraenc.h:
	* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
	(theora_enc_sink_event), (theora_enc_change_state):
	  Track initialisation state; don't try to use encoder state if we're
	  not initialised (it'll segfault).

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2007-04-18  Stefan Kost  <ensonic@users.sf.net>

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

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Allow random depths between 1 and 32 instead of only multiplies of 8.

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Set the maximum number of channels for PCM and float in the correct
	place to have it also used when creating the template caps.

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Correctly support 4, 6 and 8 channels with normal PCM and float
	wav files.

	Fix the depth and signedness calculation in extensible wav files and
	also handle 1, 2, 4, 6, 8 channels here when a file without channel
	mask is found.

	Add support for float, alaw and mulaw in extensible wav files.

	This allows correct playback of all but 5 files from
	http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
	
	(gst_riff_create_audio_template_caps):
	Add voxware and float formats to the template caps.	

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Vincent Torri <vtorri at univ-evry dot fr>

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
	Fix unused variable warning if HAVE_LOCALTIME_R is undefinied

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
	Use the correct format strings for integer formats.

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2007-04-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
	  Don't use pad_alloc_buffer_and_set_caps to create a small header
	  packet, or, worse, to create a big temporary video buffer using the
	  src pad.

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2007-04-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, buffer_probe_cb, GST_START_TEST):
	  Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.

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2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
	  streamheader_suite):
	  Add another test set up for failure

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2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/Makefile.am:
	* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
	  GST_START_TEST, streamheader_suite, main):
	  Add a test for the streamheader bug Wim fixed.

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2007-04-13  Jan Schmidt  <thaytan@mad.scientist.com>

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

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2007-04-13  Stefan Kost  <ensonic@users.sf.net>

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

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2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Try encodings from all environment variables, not just those in the
	  first environment variable that is set.

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2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_chain):
	Add some debug.

	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	Added check for videorate changing caps handling. Closes #421834.

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2007-04-12  Michael Smith  <msmith@fluendo.com>

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

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2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  API: add gst_tag_freeform_string_to_utf8() (#405072).

	* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
	  Use gst_tag_freeform_string_to_utf8() here.

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2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
	(gst_gdp_pay_sink_event):
	Make sure we set the IN_CAPS flag correctly.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Get the IN_CAPS flag before we call functions that mess with the flags.

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2007-04-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
	  gst_gdp_pay_chain, gst_gdp_pay_sink_event):
	  Only stamp buffers with offset/offset_end right before they get
	  pushed.  This ensures offset continuity, which was not the case
	  before as shown by
	  gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE

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2007-04-06  Wim Taymans  <wim@fluendo.com>

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

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2007-04-06  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-05  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving):
	Don't try to create invalid calibration parameters by making the
	internal time go backwards, instead make external time go forward.

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2007-04-05  Wim Taymans  <wim@fluendo.com>

	Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>

	* gst/playback/gstplaybasebin.c: (add_stream):
	Fix leak in add_stream(), when g_value_set_object() increases the
	refcount of streaminfo object. Fixes #426250.

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2007-04-03  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add a test pattern called "circular", which has concentric
	  rings with varying radial frequency.  The main purpose of this
	  pattern is to test fidelity loss in a filter or scaler element.
	  Notably, this pattern is scale invariant, and is optimally viewed
	  with a width (and height) of 400.

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2007-04-03  Wim Taymans  <wim@fluendo.com>

	Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>

	* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
	(deactivate_free_recursive):
	Decodebin2 doesn't unref pads it obtains in some occasions:
	- multiqueue src pads, when either connecting further or exposing
	- sink pads of new autoplugged elements
	- peer pads when recursively freeing elements
	Fixes #425455.

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2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Add audio/x-raw-float support, now that audioconvert support
	non-native endianness floats.

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2007-03-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	  gstreamer-plugins-base.pc doesn't exist, it's
	  gstreamer-plugins-base-0.10.pc.

René Stadler's avatar
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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: René Stadler <mail at renestadler dot de>
	with some minor changes

	* gst-libs/gst/floatcast/floatcast.h:
	Use more efficient float endianness conversion functions that don't
	involve 2 function calls per value.
	* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	(check_default), (audio_convert_prepare_context):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_parse_caps), (make_lossless_changes):
	Support non-native endianness floats as input and output.
	Fixes #339838.
	* tests/check/elements/audioconvert.c: (verify_convert),
	(GST_START_TEST):
	Add unit tests for the non-native endianness float conversions.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_base_init),
	(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Add Private structure.
	Bring element code to 2007.
	Parse clock-base caps param and use it when generating the
	newsegment.
	Reset variables before going to PAUSED.
	Fix some docs.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_get_adapter):
	Add RTCP docs.
	Fix some more docs.

	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
	(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
	(gst_rtcp_buffer_get_packet_count), (read_packet_header),
	(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
	(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
	(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
	(gst_rtcp_packet_sr_get_sender_info),
	(gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
	(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
	(gst_rtcp_packet_sdes_get_chunk_count),
	(gst_rtcp_packet_sdes_first_chunk),
	(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
	(gst_rtcp_packet_bye_get_ssrc_count),
	(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_get_reason_len),
	(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Add new helper object for parsing and creating RTCP messages.

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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	PCM samples with width=8 must be always unsigned, no matter what
	depth they have.

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2007-03-29  Andy Wingo  <wingo@pobox.com>

	* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
	perfect offsets also, not just timestamps.

	* tests/check/elements/videorate.c (test_more): Test that given
	any incoming offsets, that videorate produces perfect offsets.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/riff/riff-ids.h:
	Add some more RIFF formats.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	(gst_rtp_buffer_default_clock_rate):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Fix fixed payload names and docs.
	Added method to get the default clock rates of fixed payload types.
	API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()

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2007-03-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* tests/check/pipelines/.cvsignore:
	Add new vorbisdec test to cvsignore.

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2007-03-28  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
	(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
	(gst_base_audio_sink_set_property),
	(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
	(clock_convert_external), (gst_base_audio_sink_resample_slaving),
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_async_play):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	Store private stuff in GstBaseAudioSinkPrivate.
	Add configurable clock slaving modes property.
	API:: GstBaseAudioSink::slave-method property
	Some more latency reporting tweaks.
	Added skew based clock slaving correction and make it the default until
	the resampling method is more robust.

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2007-03-27  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/audioconvert.c:
	Add docs to the integer pack functions and implement proper
	rounding. Before we had rounding towards negative infinity, i.e.
	always the smaller number was taken. Now we use natural rounding,
	i.e. rounding to the nearest integer and to the one with the largest
	absolute value for X.5. The old rounding introduced some minor
	distortions. Fixes #420079
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix one unit test that assumed the old rounding and added unit tests
	for checking signed/unsigned int16 <-> signed/unsigned int16 with
	depth 8, one for signed int16 <-> unsigned int16 and one for the new
	rounding from signed int32 to signed/unsigned int16.

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2007-03-27  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
	(gst_audio_convert_transform_caps):
	  Fix typo in debug line introduced recently, as pointed out on irc.

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2007-03-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	* tests/check/libs/tag.c: (GST_START_TEST):
	  Make sure we parse floating-point numbers in vorbis comments
	  correctly with either '.' or ',' as separator, no matter what
	  the current locale is. Add unit test for this too.

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2007-03-26  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler  <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
	  When writing out floating-point numbers to vorbis comment tags, always
	  use the same character as separator no matter what the current locale is
	  (fixes #423051).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit tests for replaygain tags in vorbis comments (closes #423055).

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2007-03-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
	  vorbis_handle_data_packet):
	  Correctly set DURATION to generate a timestamp-continuous stream.
	  One bug left at the end; see
	  ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
	* tests/check/Makefile.am:
	* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
	  Add a test to check this.  Without the above patch this test fails.

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2007-03-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/rtp/Makefile.am:
	The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.

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2007-03-23  Michael Smith  <msmith@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_reset), (gst_video_rate_chain):
	  If videorate changes caps, we can no longer use the old buffer
	  (which may have a different size, incompatible with our caps).
	  So don't do that; just duplicate the new frame more times.

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2007-03-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	Remove playbin's override of the set_clock vmethod. It's irrelevant
	after Wim's commit on the 19th.

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2007-03-22  Wim Taymans  <wim@fluendo.com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
	(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
	* ext/gnomevfs/gstgnomevfssrc.h:
	Don't cache file sizes. Fixes #341078.

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2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (add_sink):
	  Use GST_PTR_FORMAT to log caps. 

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2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/subparse/samiparse.c: (handle_start_font):
	  Special-case some more colour names that pango doesn't handle by
	  default. Fixes #420578.

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	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  If we get a zero-sized input buffer, don't pass it to libvorbis, as
	  that marks EOS internally. After that, libvorbis will buffer all
	  input data, and encode none of it, eventually leading to memory
	  exhaustion.

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	* gst/playback/gstdecodebin.c: (remove_fakesink):
	Don't post STATE_DIRTY anymore.

	* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
	(gst_play_bin_change_state):
	Remove stream_time reset in seek handling, core does that now.
	Disable clocking for live pipelines by forcing a NULL clock to the
	complete pipeline, core is too smart now for our previous hack.
	We can always autoplug in PAUSED now.

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2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS:  Update this file, change the formatting to make
	it more consistent, plus more machine readable.

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2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(strip_width_64), (append_with_other_format):
	  Previous fix was too simplistic, and broke the tests. Use a better
	  approach; only strip 64 from widths for integer audio.

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2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(gst_audio_convert_transform_caps):
	  We don't support 64 bit integer audio, so don't try to claim we can.
	  Stops us producing caps don't match our template caps.
	  Update comments.

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2007-03-15  Michael Smith  <msmith@fluendo.com>

	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont), (audioresample_transform):
	  Don't trigger discontinuities for very small imperfections; a filter
	  flush will sound bad, and many plugins have rounding errors leading
	  to these.

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	Patch by Olivier Crete <olivier.crete@collabora.co.uk>

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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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	API: add "min-ptime" property to RTP base audio payloader.
	API: add gst_base_rtp_audio_payload_push().
	API: add gst_base_rtp_audio_payload_get_adapter().
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	Fixes #415001
	Indentation/whitespace/documentation fixes.

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	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	(audioresample_transform_size), (audioresample_do_output),
	(audioresample_transform), (audioresample_pushthrough): Handle
	discontinuous streams.
	* gst/audioresample/gstaudioresample.h:
	* tests/check/elements/audioresample.c:
	(test_discont_stream_instance), (GST_START_TEST),
	(audioresample_suite): Add a test for discontinuous streams.
	* win32/common/config.h: Updated.

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2007-03-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.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:
	  Update translations from translation project.

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	* gst/audioresample/debug.h:
	* gst/audioresample/resample.c: (resample_init):
	  Since I really am not interested in a debug line for each sample
	  being processed, move the library's debugging to its own category,
	  libaudioresample

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2007-03-13  Michael Smith  <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  Since the plugin doesn't support anything other than 4:2:0 right
	  now, post an error and fail if we get something else. Won't matter
	  until libtheora supports the other pixel formats, but hopefully
	  that'll be soon...

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2007-03-10  Sebastien Moutte  <sebastien@moutte.net>

	* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
	Use gst_guint64_to_gdouble for conversion.
	* win32/MANIFEST:
	Add new files to the win32 MANIFEST.
	* win32/common/libgstaudio.def:
	* win32/common/libgstpbutils.def:
	Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstplaybin.dsp:
	Change the link to libgstpbutils.lib.
	* win32/vs6/libgstdecodebin2.dsp:
	Add a new project for decodebin2.
	* win32/vs6/libgstpbutils.dsp:
	Add a new project for pbutils.

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Also accept partial dates with only year and month,
	  like 1999-12-00 (fixes #410396 even more).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit test for the above.

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	  Add unit test for MPL2 subtitle format (#413799).

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Kamil Pawlowski  <kamilpe gmail com>

	* gst/subparse/Makefile.am:
	* gst/subparse/gstsubparse.c:
	(gst_sub_parse_data_format_autodetect),
	(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
	(gst_subparse_type_find):
	* gst/subparse/gstsubparse.h:
	* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
	* gst/subparse/mpl2parse.h:
	  Add support for MPL2 subtitle format (#413799).

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We require core CVS for the new buffer metadata copy functions.

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

	* gst-libs/gst/tag/gstid3tag.c:
	Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.

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	Patch by: Alex Lancaster <alexl at users sourceforge net>

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

	* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
	(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
	Improve adapter usage and comments.

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

	* ext/pango/gsttextrender.c: (gst_text_render_chain):
	* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
	* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
	Use new metadata copy function.

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_transform):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
	Basetransform copied the metadata for us.

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
	(gst_text_overlay_video_event):
	  Some more logging. Only accept newsegment events in TIME format and
	  send a WARNING message if they are not in TIME format.

	* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
	(gst_sub_parse_chain), (gst_sub_parse_sink_event):
	* gst/subparse/gstsubparse.h:
	  No need to allocate GstSegment structure dynamically, just put it
	  into the instance structure; ignore newsegment events in BYTE
	  format and in particular don't let it overwrite our saved TIME
	  segment from the last seek.

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2007-03-09  Michael Smith  <msmith@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
	  Replace AC3 typefinder with one that isn't terrible, and actually
	  works usefully.

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2007-03-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_transform):
	  fix error category and translatable string
	  

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	  Fix up utils => pbutils here too.

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstsubparse.c: (handle_buffer):
	  Break out of loop in chain function as soon as possible if we get
	  a non-OK flow return.

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

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Unref the mixer if the state change fails too (if the
	alsa devices are inaccessible, for example)

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

	* tests/check/Makefile.am:
	Don't test libvisual elements in the states check, because libvisual
	seems to leak internally.

	Re-enable the alsa and states tests now that there's new suppressions
	in gst.supp.

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Don't leak the alsamixer we instantiated.

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

	* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state), (gst_ximagesink_reset),
	(gst_ximagesink_finalize):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
	(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
	Move some cleanup stuff from the state change handler into a _reset()
	function that can be called from _finalize(). This ensures that things
	get freed even if (for some reason) the NULL->READY state transition
	fails in the parent class.
	Even if a parent state change fails, process our downward state change
	logic instead of bailing out early.
	Free the correct xcontext pointer in ximagesink's xcontext_clear.

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

	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	Extra log line.

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
	* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
	Use pango_font_description_set_family_static instead of 
	pango_font_description_set_family to save a string copy (it was
	leaking due to the strdup anyway)

	* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
	* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
	* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
	Chain up in finalize.

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2007-03-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/interfaces/mixertrack.c:
	(gst_mixer_track_class_init), (gst_mixer_track_get_property),
	(gst_mixer_track_set_property):
	  API: add "untranslated-label" property which should be set by
	  implementations at construct time (#414645).

	* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Set "untranslated-label" when constructing mixer track objects.

	* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
	  Unit test to check the above.

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2007-03-07  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	Fix confusing debug message.

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2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-plugins-base.doap:
	update doap file with new version

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2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

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=== release 0.10.12 ===

2007-03-07  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.12, "Zombie Horde"

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2007-03-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump version to 0.10.11.4 pre-release

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	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	Fix regression that made GStreamer skip the first samples of audio.
	Fixes #414684.

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2007-03-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump version to 0.10.11.3 pre-release

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	* po/POTFILES.in:
	  Update paths for the rename from utils to pbutils to fix the build.

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2007-03-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/Makefile.am:
	  Change directory to install headers in from gst/utils to gst/pbutils
	  as well.

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2007-03-05  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/descriptions.c:
	(gst_pb_utils_get_source_description),
	(gst_pb_utils_get_sink_description),
	(gst_pb_utils_get_decoder_description),
	(gst_pb_utils_get_encoder_description),
	(gst_pb_utils_get_element_description),
	(gst_pb_utils_add_codec_description_to_tag_list),
	(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
	* gst-libs/gst/pbutils/descriptions.h:
	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/pbutils/missing-plugins.c:
	(gst_missing_uri_source_message_new),
	(gst_missing_uri_sink_message_new),
	(gst_missing_element_message_new),
	(gst_missing_decoder_message_new),
	(gst_missing_encoder_message_new),
	(gst_missing_plugin_message_get_description):
	* gst-libs/gst/pbutils/missing-plugins.h:
	* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
	* gst-libs/gst/pbutils/pbutils.h:
	* gst-libs/gst/utils/Makefile.am:
	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/base-utils.h:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/descriptions.h:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/install-plugins.h:
	* gst-libs/gst/utils/missing-plugins.c:
	* gst-libs/gst/utils/missing-plugins.h:
	* gst-plugins-base.spec.in:
	* gst/playback/Makefile.am:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybasebin.c: (setup_subtitle),
	(gen_source_element):
	* gst/playback/gstplaybin.c: (plugin_init):
	* tests/check/Makefile.am:
	* tests/check/libs/pbutils.c: (GST_START_TEST),
	(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
	* tests/check/libs/utils.c:
	  rename utils to pbutils

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2007-03-02  Jan Schmidt  <thaytan@mad.scientist.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
	Add documentation for decodebin2 that indicates that the API
	is still unstable.

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2007-03-01  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Update to 0.10.11.2 (0.10.12 pre-release)

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	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	base time is irrelevant here.

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2007-03-01  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	Improve debugging.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_query), (gst_base_audio_sink_event),
	(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
	Improve latency and clock slaving calculations.
	Improve slave clock calibration.

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_commit_full):
	When we are asked to render N sample to 0 bytes, return N.

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2007-03-01  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	(gst_alsasink_write), (gst_alsasink_reset):
	* ext/alsa/gstalsasink.h:
	Remove unused dispose function.
	Rename lock to not interfere with alsasrc lock.

	* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
	(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
	(gst_alsasrc_read), (gst_alsasrc_reset):
	* ext/alsa/gstalsasrc.h:
	Implement finalize function.
	Use lock to protect alsa access.
	Implement _reset.
	Fine tune sw params.

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2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  Convert to new AG_GST style.

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2007-02-28  Wim Taymans  <wim@fluendo.com>

	Patch by: Ed Catmur <ed at catmur dot co dot uk>

	* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
	(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
	Fix race condition when rapidly switching visualisations in playbin.
	Fixes #401029.

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	* tests/check/Makefile.am:
	Include local stuff before system installed things in LDFLAGS and
	CFLAGS.

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	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
	Improve debugging.

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	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
	(gst_v4lsrc_fixate), (gst_v4lsrc_query):
	* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
	Fix duration and timestamping, taking latency into account.
	Implement latency query.

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	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	(gst_audio_clock_new):
	Fix clock name.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_init), (gst_base_audio_sink_query):
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
	(gst_base_audio_src_create):
	Improve latency query code.
	Use proper clock names.

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	* tests/check/generic/states.c: (GST_START_TEST):
	  Copy the states.c test from core again
	* tests/check/Makefile.am:
	  ignore cdio and cdparanoiasrc

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	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index), (check_default),
	(audio_convert_prepare_context), (audio_convert_convert):
	  Also make valgrind happy and avoid copying data in some cases.

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2007-02-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
	(gst_audio_convert_transform_caps):
	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Don't run inplace if that overwrites source data as we go. Add more
	  tests. Fixes #339837 even more.

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	* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
	(msg_segment_done): Fix various seeking bugs (Slider was not
	updating when doing a non flushing seek, Reverse playback 
	on segment seek was wrong).

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	* tests/examples/seek/seek.c: (stop_seek):
	When we stop scrubbing, don't leave the pipeline PLAYING when we
	requested a PAUSED state.

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2007-02-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse date strings in vorbis comments that have an invalid (zero)
	  month or day (#410396).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Test case for the above.

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2007-02-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Loïc Minier <lool+gnome at via ecp fr>

	* configure.ac:
	* ext/alsa/Makefile.am:
	* gst/audiotestsrc/Makefile.am:
	  Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).

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2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: point out that the application needs to assist playbin
	  with buffering.

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2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	* tests/check/libs/utils.c: (missing_msg_check_getters):
	  Change GStreamer marker prefix in detail string from 'gstreamer.net'
	  to just 'gstreamer'. Document the caps string component of the
	  decoder/encoder detail a bit better, since not everyone will be
	  familiar with the GStreamer media type/caps system (but they better
	  enjoy nested itemized lists).

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	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
	  Fix copying of GstNetBuffer (would crash before, or at least lead to
	  invalid memory access, #410772), for now by copying the GstBuffer copy
	  code from the core over here so we can copy the GstBuffer fields on a
	  provided buffer instance (of type GstNetBuffer in this case). Would be
	  better to fix this with some support by the core though (and in the long
	  run change the broken GstBuffer/GstMiniObject copy semantics, #393099).

	* tests/check/Makefile.am:
	  Enable unit test for GstNetBuffer.

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	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_init): Disable pull-mode activation until we
	figure out how to make audio sinks go to PLAYING.

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	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
	(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
	* gst/audioconvert/gstchannelmix.h:
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Add float as an intermediate format, as well as float mixing. Enable
	  test that was failing before. Fixes #339837

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	* tests/examples/seek/seek.c: (do_seek):
	Undo the previous commit: -1 as a stop time implies that the stop
	time is the end of file, clearing any previously configured segment.

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	* tests/examples/seek/seek.c: (do_seek):
	Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.

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	* gst/volume/gstvolume.c: (volume_process_int16),
	(volume_process_int16_clamp), (volume_set_caps):
	  Unbreak volume, value remains gint.

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	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_set_volume),
	(gst_volume_init), (volume_process_double), (volume_process_float),
	(volume_process_int16), (volume_process_int16_clamp),
	(volume_set_caps), (volume_transform_ip), (volume_update_volume):
	* gst/volume/gstvolume.h:
	  Extend float audio support (double) and some int->uint cleanups.

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	* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
	(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
	(sort_end_pads), (gst_decode_group_expose),
	(gst_decode_group_hide):
	Don't free groups from the streaming threads. Just put them aside and
	free them in dispose.

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	* gst/playback/gstdecodebin2.c: (connect_element),
	(pad_added_group_cb), (gst_decode_group_check_if_blocked),
	(sort_end_pads), (gst_decode_group_expose):
	Handle dynamic pads within groups.
	Sort pads before exposing them in order to make playbin happy.
	There still is a race with the multiqueue filling up. This should be
	solved separately.
	Fixes #398721

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	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	  Some more docs (and descriptions for two subtitle formats).

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	* gst-libs/gst/audio/audio.c:
	  Fix documentation.

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	Patch by: Yves Lefebvre  <ivanohe abacom com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
	  Don't leak caps. Fixes #408278.

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	* ext/cdparanoia/gstcdparanoiasrc.h:
	* ext/ogg/gstoggdemux.h:
	* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
	(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
	(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/interfaces/videoorientation.h:
	* gst/adder/gstadder.h:
	  More docs coverage and some ChangeLog surgery (add missing names)

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2007-02-15  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c:
	(gst_ximagesink_calculate_pixel_aspect_ratio):
	* sys/xvimage/xvimagesink.c:
	(gst_xvimagesink_calculate_pixel_aspect_ratio):
	Small constifications.

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2007-02-15  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
	(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
	(gst_base_audio_sink_async_play),
	(gst_base_audio_sink_change_state):
	Answer latency query.
	Use configured latency when syncing.
	Fix clock slaving.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
	(gst_base_audio_src_query), (gst_base_audio_src_change_state):
	Fix possible memleak.
	Implement latency query.
	Small cleanups.

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2007-02-15  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
	Ignore errors in reset, these are not fatal. They also grab the element
	lock which is already taking when this function is called. Fixes
	#405451.

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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Remove 'tests/examples/xerror/Makefile' from output files again.

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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Also crossref against gst-plugins-base-libs.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.

	* gst-libs/gst/audio/audio.h:
	  Source formatting.

	* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
	  Add own debug category.

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
	  (#403597).

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2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (setup_source):
	  When we have external subtitles and wait for the subtitle decodebin
	  to get up and running, we set up a (sync) bus handler for the
	  subtitle decodebin, so we can stop waiting when it posts an error
	  message. However, we should do that before we set the subtitle
	  decodebin's state to playing, otherwise things are racy and we might
	  miss error messages posted before we had a chance to set up the bus.
	  This should finally fix totem hanging on .txt pseudo-subtitle files.
	  
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2007-02-10  Sébastien Moutte  <sebastien at moutte dot net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	  Use gst_gdouble_to_guint64 for conversions.
	* win32/common/config.h.in:
	  Add a define for GST_INSTALL_PLUGINS_HELPER
	* win32/common/libgstaudio.def:
	* win32/common/libgstcdda.def:
	* win32/common/libgstnetbuffer.def:
	* win32/common/libgstrtp.def:
	* win32/common/libgutils.def:
	  Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstnetbuffer.dsp:
	* win32/vs6/libgstplaybin.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstvorbis.dsp:
	* win32/vs6/libgstcdda.dsp:
	* win32/vs6/libgstgdp.dsp:
	* win32/vs6/libgstutils.dsp:
	  Update and add new project files.

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2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
	(subrip_remove_unhandled_tags), (parse_subrip):
	  For SubRip (.srt) subtitles, ignore all markup tags we don't
	  handle (like font tags, for example).

	* tests/check/elements/subparse.c:
	  Add test for this.

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2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (add_fakesink),
	(gst_decode_bin_change_state):
	* gst/playback/gstdecodebin2.c: (add_fakesink),
	(gst_decode_bin_change_state):
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	  Don't error out if there is no fakesink in the NULL to READY state
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	  change, since when decodebin is re-used, we're only adding the
	  fakesink element in READY to PAUSED.

	* tests/check/elements/decodebin.c:
	(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
	(decodebin_suite):
	  Minimal unit test to make sure we can use the same decodebin
	  instance twice (at least with audiotestsrc input).

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2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
	  Try to get devic-name from device string first, and from handle only
	  as fallback (seems to yield better results and is more robust
	  against buggy probing code on the application side).

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2007-02-08  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Julien Puydt <julien.puydt at laposte net>

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
	(gst_alsa_find_device_name):
	* ext/alsa/gstalsa.h:
	* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
	  Improve device-name detection a bit, especially in the case where
	  the device is not actually open (#405020, #405024). Move common code
	  into gstalsa.c instead of duplicating it.

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2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c:
	  Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.

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2007-02-06  Julien MOUTTE  <julien@moutte.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_clear),
	(gst_xvimagesink_interface_supported),
	(gst_xvimagesink_probe_get_properties),
	(gst_xvimagesink_probe_probe_property),
	(gst_xvimagesink_probe_needs_probe),
	(gst_xvimagesink_probe_get_values),
	(gst_xvimagesink_property_probe_interface_init),
	(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init),
	(gst_xvimagesink_get_type):
	* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
	for XVAdaptors so that one can choose the adaptor to use with 
	gstreamer-properties.

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2007-02-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	  Also mention that a conversion from double to float is suboptimal still.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstaudiofilter.c:
	(gst_audio_filter_class_init), (gst_audio_filter_change_state):
	  Clear our formats structure and free the caps contained in it when
	  shutting down.

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2007-02-05  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_callback): Update basesink->offset so that we
	pull monotonically increasing offsets instead of, um, seeking back
	to 0 each time. Fixes alsasrc ! alsasink!

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2007-02-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videoscale/gstvideoscale.c:
	  A width and height of 1 makes us crash, so increase minimum size to
	  2x2 pixels until someone feels like fixing this (#404512).

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2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
	  Add small test to make sure request pads are cleaned up properly
	  even if oggmux never changes state out of NULL.

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2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/utils.c: (GST_START_TEST):
	  Fix unit test. Turns out things work much better when you
	  NULL-terminate string arrays. Should make p5 build bot happy again.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	(gst_audio_filter_template_base_init),
	(gst_audio_filter_template_class_init),
	(gst_audio_filter_template_init),
	(gst_audio_filter_template_set_property),
	(gst_audio_filter_template_get_property),
	(gst_audio_filter_template_setup),
	(gst_audio_filter_template_filter),
	(gst_audio_filter_template_filter_inplace), (plugin_init):
	  Oops, forgot to commit fixed-up example.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
	(gst_audio_filter_class_init), (gst_audio_filter_init),
	(gst_audio_filter_set_caps),
	(gst_audio_filter_class_add_pad_templates):
	* gst-libs/gst/audio/gstaudiofilter.h:
	  Port GstAudioFilter to 0.10. This change technically breaks
	  API and ABI (and thus also every library developer's heart),
	  but seems justifiable on the grounds that the base class was
	  completely unusable before (ie. would