=== release 0.9.2 ===

2005-09-06  Thomas Vander Stichele  <thomas at apestaart dot org>

	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/random/ChangeLog-0.8:
	  releasing 0.9.2, "Spoon"

2005-09-05  Michael Smith <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init):
	  libvorbis accepts quality as low as -0.1, not just 0.0. So accept
	  that in the vorbisenc element.

2005-09-04  Thomas Vander Stichele  <thomas at apestaart dot org>

	* common/gtk-doc-plugins.mak:
	* docs/plugins/Makefile.am:
	  fix distcheck
	* gst/audioresample/resample.c:
	  fix wrong docstring

2005-09-04  Thomas Vander Stichele  <thomas at apestaart dot org>

	* common/gst-xmlinspect.py:
	* common/gtk-doc-plugins.mak:
	  only inspect plugins for this given package
	  require gst-python 0.9

2005-09-03  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	* autogen.sh:
	* common/gst-xmlinspect.py:
	* configure.ac:
	* docs/Makefile.am:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/tmpl/element-gnomevfssink.sgml:
	* docs/plugins/tmpl/element-multifdsink.sgml:
	* docs/plugins/tmpl/element-tcpserversink.sgml:
	* docs/plugins/tmpl/element-vorbisenc.sgml:
	* gst-plugins-base.spec.in:
	  various doc-related updates

2005-08-31  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Resync if the buffer timestamps drift more than a 10th 
	of a second.

2005-08-31  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_property),
	(gst_v4lsrc_get_property):
	  The 'timestamp-offset' property is registered as an int64, so
	  let's use g_value_{set|get}_int64() in our setter and getter
	  functions (makes it work and fixes warnings with gst-inspect).

2005-08-30  Wim Taymans  <wim@fluendo.com>

	* check/elements/audioconvert.c: (setup_audioconvert):
	* check/elements/audioresample.c: (setup_audioresample):
	* check/elements/volume.c: (setup_volume):
	Fix checks.

2005-08-30  Thomas Vander Stichele  <thomas at apestaart dot org>

	* common/gtk-doc-plugins.mak:
	* common/plugins.xsl:
	* docs/plugins/Makefile.am:
	  make module a param

2005-08-30  Stefan Kost  <ensonic@users.sf.net>

	* examples/seeking/seek.c: (make_mp3_pipeline),
	(make_mpeg_pipeline), (seek_cb), (start_seek), (stop_seek),
	(play_cb), (pause_cb), (stop_cb):
	  update the example

2005-08-30  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c: (gst_volume_class_init),
	(volume_transform):
	  do not update controlled params, if buffer has no timestamp

2005-08-29  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/sine/Makefile.am:
	* gst/volume/Makefile.am:
	  controllerized elements also need to link against controller-libs ;)

2005-08-29  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstgconf.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
	(gst_sinesrc_create):
	* gst/volume/gstvolume.c: (gst_volume_class_init),
	(volume_transform):
	  controllerized two audio plugins

2005-08-29  Andy Wingo  <wingo@pobox.com>

	* ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push) 
	(vorbis_handle_data_packet): Fix some int overflow errors.

	* ext/ogg/gstoggdemux.c (gst_ogg_demux_init): Init total_time to
	-1.
	(gst_ogg_demux_perform_seek): Clamp segment_stop only if it's
	valid.
	(gst_ogg_pad_submit_packet): Subtract the chain's begin_time only
	if it's valid. Fixed streaming-mode playback.

	* check/elements/volume.c (cleanup_volume): Fix for running
	CK_FORK=no.

	* check/elements/audioconvert.c: Convert from native endian, not
	little endian.

2005-08-29  Michael Smith <msmith@fluendo.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogg.c: (plugin_init):
	* ext/ogg/gstoggparse.c: (gst_ogg_parse_get_type), (free_stream),
	(gst_ogg_parse_delete_all_streams), (gst_ogg_parse_new_stream),
	(gst_ogg_parse_find_stream), (gst_ogg_parse_base_init),
	(gst_ogg_parse_class_init), (gst_ogg_parse_init),
	(gst_ogg_parse_dispose), (gst_ogg_parse_submit_buffer),
	(gst_ogg_parse_append_header), (gst_ogg_parse_is_header),
	(gst_ogg_parse_buffer_from_page), (gst_ogg_parse_chain),
	(gst_ogg_parse_change_state), (gst_ogg_parse_plugin_init):
	Add an ogg parser element.

2005-08-28  Andy Wingo  <wingo@pobox.com>

	* Updates for two-arg init from GST_BOILERPLATE_FULL.

2005-08-26  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/audioconvert.c: (if), (float),
	(audio_convert_get_func_index), (check_default),
	(audio_convert_clean_fmt), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_get_sizes),
	(audio_convert_convert):
	Cleanups.

2005-08-26  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/audioconvert.c: (if), (float),
	(audio_convert_get_func_index), (check_default),
	(audio_convert_clean_fmt), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_get_sizes),
	(audio_convert_convert):
	More elegant and working temp buffer selection algo.

2005-08-26  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/audioconvert.c: (if), (float),
	(audio_convert_get_func_index), (check_default),
	(audio_convert_clean_fmt), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_get_sizes),
	(get_temp_buffer), (audio_convert_convert):
	Use realloc else we lose our original data.

2005-08-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioresample/gstaudioresample.c:
	  use base class' newsegment to properly timestamp

2005-08-26  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/audioconvert.c: (if), (float),
	(audio_convert_get_func_index), (check_default),
	(audio_convert_clean_fmt), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_get_sizes),
	(get_temp_buffer), (audio_convert_convert):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
	(gst_audio_convert_transform_caps),
	(gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
	Oops, allocate enough space to perform the channel mix.

2005-08-26  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c: (if), (float),
	(audio_convert_get_func_index), (check_default),
	(audio_convert_clean_fmt), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_get_sizes),
	(get_temp_buffer), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_class_init), (gst_audio_convert_init),
	(gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
	(gst_audio_convert_get_unit_size),
	(gst_audio_convert_transform_caps),
	(gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
	(gst_audio_convert_transform_ip), (gst_audio_convert_transform):
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
	(gst_channel_mix_fill_identical),
	(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
	(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
	(gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
	(gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
	(gst_channel_mix_mix):
	* gst/audioconvert/gstchannelmix.h:
	Cleanups, librarify a bit, optimize, better negotiation and more.

2005-08-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/ogg/gstoggdemux.c: (ogg_find_peek):
	Another from MikeS:
	During typefinding, don't support negative offsets
	(offsets from the end of the stream) in our typefind->peek() function
	- nothing embedded in ogg ever needs them. However, we need to recognise
	those requests and reject them, otherwise we return invalid pointers.

2005-08-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
	(vorbisdec_finalize), (vorbis_handle_type_packet):
	  Big shout-out to MikeS for fixing this giant memory leak.
	  Huzzah!

2005-08-25  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose),
	(audio_convert_get_unit_size):
	  plug some leaks

2005-08-25  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/Makefile.am:
	* check/elements/audioconvert.c: (setup_audioconvert),
	(cleanup_audioconvert), (get_int_caps), (verify_convert),
	(GST_START_TEST), (audioconvert_suite), (main):
	  add a test for audioconvert
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/gstaudioresample.h:
	  set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
	  note that for buffers of 1/3 sec this means DURATION(c) is 
	  one nanosecond more than for a and b

2005-08-25  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/Makefile.am:
	* check/elements/audioresample.c: (setup_audioresample),
	(cleanup_audioresample), (fail_unless_perfect_stream),
	(test_perfect_stream_instance), (GST_START_TEST),
	  add a check for audioresample
	(audioresample_suite), (main):
	* check/elements/volume.c: (GST_START_TEST):
	  remove unused method
	* gst/audioresample/gstaudioresample.c:
	  set correct buffer parameters since we're changing them
	* gst/audioresample/resample_ref.c: (resample_scale_ref):
	  add some debug

2005-08-25  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioresample/debug.c:
	* gst/audioresample/gstaudioresample.c:
	  add room for extra overlap samples when asked to transform size
	  protect against possible mem corruption and check for discrepancies
	  between written size and outbuffer's size so we can warn for
	  potential problems
	* gst/audioresample/resample.c: (resample_init),
	(resample_get_output_size_for_input), (resample_get_output_size),
	(resample_set_n_channels), (resample_set_format):
	  set debug level based on RESAMPLE_DEBUG env var
	  make sure that get_output_size* returns a whole number of
	  sample_size
	  set sample_size each time either channel or format is set
	* gst/audioresample/resample_chunk.c: (resample_scale_chunk):
	* gst/audioresample/resample_functable.c:
	(resample_scale_functable):
	* gst/audioresample/resample_ref.c: (resample_scale_ref):
	  remove r->sample_size, it's done in resample.c now
	  add some debugging to the ref implementation
	  make sure we only give back bytes that are wholes of the sample
	  size

2005-08-25  Jan Schmidt  <thaytan@mad.scientist.com>
	* gst/playback/gstplaybasebin.c: (fill_buffer):
	Revert unpopular change for GST_MESSAGE_SRC to GObject.

2005-08-25  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  made set_caps function static

2005-08-24  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
	(gst_vorbisenc_change_state):
	Stop leaking taglists.

2005-08-24  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
	(gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
	(gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
	(gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
	(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
	Parse seeking events better.
	Unref static caps.
	Generate correct newsegment events, fixes seeking in live oggs.

	* ext/theora/theoradec.c: (theora_dec_src_query),
	(theora_dec_src_event), (theora_dec_src_getcaps),
	(theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
	Use newsegment values to report correct play time.

	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	(vorbis_dec_src_event), (vorbis_dec_sink_event):
	* ext/vorbis/vorbisdec.h:
	Parse and use newsegment values to report correct play time.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_event), (gst_base_audio_sink_render):
	Clear ringbuffer on flush.
	Use newsegment values to calculate playback time.

	* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
	Basesink does newsegment calculations for us now.

2005-08-24  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/Makefile.am:
	* configure.ac:
	  add core's plugins to the mix so that playbin works
	* check/generic/states.c: (GST_START_TEST):
	  set a 0 timeout on pipelines, so they don't force the next
	  state change
	* gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
	(gst_play_base_bin_change_state):
	  remove the crappy error handling and do GST error handling

2005-08-24  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/Makefile.am:
	* check/generic/states.c: (GST_START_TEST), (states_suite), (main):
	  add same test as to core, it bitches out on playbin atm.

2005-08-24  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	Remove audioscale.

2005-08-24  Wim Taymans  <wim@fluendo.com>

	* gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
	(gst_videoscale_prepare_size), (parse_caps),
	(gst_videoscale_set_caps), (gst_videoscale_get_size),
	(gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
	(gst_videoscale_transform):
	* gst/videoscale/gstvideoscale.h:
	Refactor, make use of BaseTranform really well.

2005-08-24  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  compile audioresample
	* gst/audioresample/Makefile.am:
	* gst/audioresample/buffer.c:
	* gst/audioresample/functable.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/gstaudioresample.h:
	* gst/audioresample/resample.c:
	(resample_get_output_size_for_input):
	* gst/audioresample/resample.h:
	* gst/audioresample/resample_chunk.c:
	* gst/audioresample/resample_functable.c:
	* gst/audioresample/resample_ref.c:
	  port to use basetransform; doesn't work in all cases yet

2005-08-24  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_class_init), (gst_audio_convert_init),
	(audio_convert_get_unit_size), (audio_convert_transform_caps),
	(audio_convert_fixate_caps), (audio_convert_set_caps),
	(audio_convert_transform),
	(gst_audio_convert_buffer_to_default_format),
	(gst_audio_convert_buffer_from_default_format),
	(gst_audio_convert_channels):
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  port to basetransform
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_class_init),
	(gst_ffmpegcsp_get_unit_size):
	* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
	(gst_videoscale_transform_caps), (gst_videoscale_get_unit_size):
	  fix for basetransform changes

2005-08-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* check/Makefile.am:
	  Add CHECK_CFLAGS and LDFLAGS

	* gst/playback/gstplaybasebin.c: (fill_buffer):
	  GST_MESSAGE_SRC became a GObject

2005-08-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
	(gst_ring_buffer_clear_all):
	* gst-libs/gst/audio/gstringbuffer.h:
	Added function to clear the ringbuffer.

2005-08-24  Andy Wingo  <wingo@pobox.com>

	* sys/v4l/gstv4lelement.c (gst_v4lelement_start) 
	(gst_v4lelement_stop): Call _start and _stop for xoverlay instead
	of _open and _close.

	* sys/v4l/gstv4lxoverlay.h:
	* sys/v4l/gstv4lxoverlay.c (gst_v4l_xoverlay_set_xwindow_id): Open
	an Xv connection here, instead of all the time. Make Xv only be
	loaded if you axe for it. Kindof a workaround for buggy behaviour
	of Xv when using remote xservers (XvQueryExtension would block).
	(gst_v4l_xoverlay_stop, gst_v4l_xoverlay_start): New functions,
	replace the _open and _close public API. Only start the xv
	connection if necessary.
	(gst_v4l_xoverlay_open, gst_v4l_xoverlay_close): Made static.

2005-08-23  David Schleef  <ds@schleef.org>

	* gst/audioresample/Makefile.am: Leet audioresampling code
	* gst/audioresample/buffer.c:
	* gst/audioresample/buffer.h:
	* gst/audioresample/debug.c:
	* gst/audioresample/debug.h:
	* gst/audioresample/functable.c:
	* gst/audioresample/functable.h:
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/gstaudioresample.h:
	* gst/audioresample/resample.c:
	* gst/audioresample/resample.h:
	* gst/audioresample/resample_chunk.c:
	* gst/audioresample/resample_functable.c:
	* gst/audioresample/resample_ref.c:

2005-08-23  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_vorbis_pipeline),
	(make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
	Small seek updates.

2005-08-23  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c
	(gst_base_audio_src_fixate): Only fixate endianness if it is
	present in the caps.

2005-08-22  Andy Wingo  <wingo@pobox.com>

	* ext/alsa/gstalsasink.c (gst_alsasink_get_property): 
	* ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
	device-name property.

	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
	close_device in the ring buffer, like gstaudiosink.

	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
	macro to implement the interface without much code. Cleanups. 

	* ext/alsa/gstalsasrc.h:
	* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
	READY.

	* ext/alsa/Makefile.am: Add new files.
	* ext/alsa/gstalsamixerelement.c: 
	* ext/alsa/gstalsamixerelement.c: Split element code out from
	mixer code so that alsasrc can be a mixer too.

2005-08-21  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/elements/volume.c: (setup_volume), (cleanup_volume),
	(GST_START_TEST):
	* check/elements/vorbisdec.c: (setup_vorbisdec),
	(cleanup_vorbisdec), (GST_START_TEST), (vorbisdec_suite):
	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	(vorbis_handle_identification_packet),
	(vorbis_handle_comment_packet), (vorbis_handle_type_packet),
	(vorbis_handle_header_packet), (vorbis_dec_push),
	(vorbis_dec_chain):
	  use the setup/teardown methods to save code.  save code is good.

2005-08-20  Thomas Vander Stichele  <thomas at apestaart dot org>

	* check/Makefile.am:
	  add ext dir for plugins
	  add vorbisdec test conditionally
	* check/elements/volume.c: (setup_volume), (cleanup_volume),
	(GST_START_TEST), (volume_suite):
	  add a test with wrong caps
	* check/elements/vorbisdec.c: (chain_func), (setup_vorbisdec),
	(cleanup_vorbisdec), (GST_START_TEST), (vorbisdec_suite), (main):
	  add a vorbisdec test
	* ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream),
	(gst_ogg_demux_chain), (gst_ogg_demux_loop):
	  clean up debug output
	* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
	  yay, fix a segfault/security issue in vorbisdec
	  gst-launch fakesrc ! vorbisdec wasn't happy
	* ext/vorbis/vorbisenc.c: (vorbisenc_get_type),
	(gst_vorbisenc_class_init), (gst_vorbisenc_sink_setcaps),
	(gst_vorbisenc_convert_src), (gst_vorbisenc_convert_sink),
	(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
	(gst_vorbisenc_init), (gst_vorbisenc_metadata_set1),
	(gst_vorbisenc_set_metadata), (get_constraints_string),
	(update_start_message), (gst_vorbisenc_setup),
	(gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
	(gst_vorbisenc_push_packet), (gst_vorbisenc_sink_event),
	(gst_vorbisenc_chain), (gst_vorbisenc_get_property),
	(gst_vorbisenc_set_property), (gst_vorbisenc_change_state):
	* ext/vorbis/vorbisenc.h:
	  march in line
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_transform):
	  have the kow come home
	* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
	  debug my func ptr
	* gst/volume/gstvolume.c: (volume_set_caps):
	  add a debug

2005-08-20  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	* check/.cvsignore:
	* check/Makefile.am:
	* check/elements/.cvsignore:
	* check/elements/volume.c: (chain_func), (event_func),
	(setup_volume), (cleanup_volume), (GST_START_TEST), (volume_suite),
	(main):
	* configure.ac:
	  add unit test structure for gst-plugins-base
	  add a test for volume
	* gst/volume/gstvolume.c: (gst_volume_list_tracks),
	(gst_volume_set_volume), (gst_volume_get_volume),
	(gst_volume_set_mute), (gst_volume_class_init), (gst_volume_init),
	(volume_funcfind), (volume_process_float), (volume_process_int16),
	(volume_set_caps), (volume_transform), (volume_update_mute),
	(volume_update_volume), (volume_set_property),
	(volume_get_property):
	  document a little; use basetransform vmethod _set_caps

2005-08-19  Andy Wingo  <wingo@pobox.com>

	* ext/alsa/gstalsamixertrack.h:
	* ext/alsa/gstalsamixertrack.c:
	* ext/alsa/gstalsamixeroptions.h:
	* ext/alsa/gstalsamixeroptions.c:
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixer.c: Port to 0.9.

	* ext/alsa/Makefile.am: Build mixer, mixeroptions, mixertracks.
	Remove gstalsa.c and alsaclock. No more cruft here.
	
2005-08-18  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_add_to_queue),
	(gst_base_rtp_depayload_push),
	(gst_base_rtp_depayload_queue_release):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Fix for RTPBuffer changes.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data),
	(gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data),
	(gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len),
	(gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len),
	(gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data),
	(gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len),
	(gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version),
	(gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding),
	(gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to),
	(gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension),
	(gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc),
	(gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc),
	(gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker),
	(gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type),
	(gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq),
	(gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp),
	(gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len),
	(gst_rtpbuffer_get_payload):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Don't subclass GstBuffer but add methods and helper functions
	to construct and manipulate RTP packets in regular GstBuffers.

2005-08-18  Stefan Kost  <ensonic@users.sf.net>

	* gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
	  moved statement below switch
	* gst/volume/gstvolume.c: (gst_volume_class_init):
	  added debug ptr

2005-08-16  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_change_state):
	Open and close device in READY<->NULL state change.

2005-08-16  Andy Wingo  <wingo@pobox.com>

	* examples/seeking/Makefile.am: Don't compile non-compiling
	compiled objects with the compiler.

	* examples/seeking/seek.c (make_dv_pipeline): Update for new DV
	elements.

2005-08-12  Philippe Khalaf <burger@speedy.org>
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Made a thread to release the queue.
	  Removed timestamp conversion for now.

2005-08-10  Philippe Khalaf <burger@speedy.org>
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Added rtp timestamp -> gst timestamp conversion.
	  Fixed several problems with queue.

2005-08-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/net/gstnetbuffer.h:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	  Add padding (you will need to rebuild gst-plugins-base,
	  gst-plugins and all applications afterwards!)

2005-08-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
	(gst_riff_parse_chunk):
	  Fix bug in debug message and add some more debug messages.

2005-08-08  Edward Hervey  <edward@fluendo.com>

	* gst-libs/gst/riff/riff-media.c:
	backported updates since branch

2005-08-08  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_change_state): Open the device in NULL->READY
	like good elements should. Close on READY->NULL too.

	* gst-libs/gst/audio/gstaudiosink.c
	(gst_audioringbuffer_open_device,
	(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
	(gst_audioringbuffer_release): Updates for new ring buffer API,
	hook into the new audio sink api.

	* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
	(GstAudioSinkClass.close): Just open and close the device -- no
	resource allocation or configuration.
	(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
	vmethods, handle device setup and resource allocation.

	* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
	(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
	base class API.

	* gst-libs/gst/audio/gstringbuffer.h
	(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
	New vmethods.

	* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
	(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
	New API functions. The device should be opened before acquiring
	and closed after releasing.

2005-08-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/interfaces/mixer.h:
	  Reset padding to GST_PADDING.

2005-08-08  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybin.c: (remove_sinks):
	  Remove visualization from parent explicitely; works around some
	  apparent refcount issue that I haven't tracked down yet.

2005-08-08  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/alsa/gstalsasink.c: (set_hwparams):
	  Assign debug category, add negotiation debug msgs.

2005-08-07  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_start):
	  Fix error code for file-not-found to NOT_FOUND.

2005-08-05  Thomas Vander Stichele  <thomas at apestaart dot org>

	* common/gtk-doc-plugins.mak:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	  renamed to actual element names, so much nicer to look at
	* docs/plugins/tmpl/gstmultifdsink.sgml:
	  remove
	* docs/plugins/tmpl/multifdsink.sgml:
	* docs/plugins/tmpl/tcpserversink.sgml:
	  add
	* ext/alsa/gstalsa.c:
	* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
	* ext/ogg/gstoggmux.c:
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
	* gst/playback/gstdecodebin.c:
	* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
	* gst/tcp/gsttcpserversink.c:
	  various fixes and documentation additions

2005-08-05  Thomas Vander Stichele  <thomas at apestaart dot org>

	* common/Makefile.am:
	* common/gstdoc-scangobj:
	* common/gtk-doc-plugins.mak:
	* common/gtk-doc.mak:
	  add a custom scangobj that uses the registry
	  add a custom gtk-doc-plugins.mak that uses it
	  some doc build fixes
	* configure.ac:
	* docs/Makefile.am:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.types:
	* docs/plugins/tmpl/gstmultifdsink.sgml:
	  add docs for one element, multifdsink
	* gst/adder/gstadder.h:
	* gst/volume/gstvolume.h:
	  don't privatize enum
	* gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
	* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
	(gst_sync_method_get_type), (gst_client_status_get_type),
	(gst_multifdsink_class_init),
	(gst_multifdsink_client_queue_buffer),
	(gst_multifdsink_handle_client_write):
	* gst/tcp/gstmultifdsink.h:
	* gst/tcp/gsttcp.h:
	* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
	(gst_tcpclientsink_init), (gst_tcpclientsink_setcaps),
	(gst_tcpclientsink_render):
	* gst/tcp/gsttcpclientsink.h:
	* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
	(gst_tcpclientsrc_init), (gst_tcpclientsrc_create),
	(gst_tcpclientsrc_start):
	* gst/tcp/gsttcpclientsrc.h:
	* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
	(gst_tcpserversrc_init), (gst_tcpserversrc_create):
	* gst/tcp/gsttcpserversrc.h:
	* gst/typefind/gsttypefindfunctions.c:
	  remove superfluous Type stuff

2005-08-05  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybin.c: (gen_video_element):
	  Enable videoscale.

2005-08-05  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst-libs/gst/gconf/gconf.c:
	* gst-libs/gst/gconf/gconf.h:
	  Fix some Andy Problem [tm].

2005-08-04  Andy Wingo  <wingo@pobox.com>

	* gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c
	(gst_ffmpegcsp_get_size): Adapt to API changes.

	* gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
	Implement an in-place do-nothing transform.

2005-08-04  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
	(gst_ximagesink_renegotiate_size):
	  Do not set new window sizes yet if we prepare a new buffer size
	  for upstream renegotiation (software scaling) at some point in the
	  future, because this new size waqs not actually accepted yet. Once
	  accepted, renegotiation later on will set the new sizes just fine.
	  Fixes a videotestsrc ! queue ! videoscale ! ximagesink xoverlay
	  embedding testcase.

2005-08-03  Andy Wingo  <wingo@pobox.com>

	* sys/ximage/ximagesink.c (gst_ximagesink_renegotiate_size):
	(gst_ximagesink_buffer_alloc): 
	Protect the height, width, and desired_caps with the pool_lock.
	Fixes videotestsrc ! queue ! ximagesink.

2005-08-02  Edward Hervey  <edward@fluendo.com>

	* gst/volume/gstvolume.c:
	include left from controller cleanup

2005-08-02  Jan Schmidt  <thaytan@mad.scientist.com>
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_change_state):
	  Stop collectpads before calling the parent state
	  change function on PAUSED->READY.

2005-08-01  Jan Schmidt  <thaytan@mad.scientist.com>
	* configure.ac:
	  When testing for X libs, use the X CFlags 
	* gst/adder/gstadder.c: (gst_adder_change_state):
	  Stop the collectpads before calling parent state change function
	  on PAUSED->READY, otherwise we deadlock deactivating pads.

2005-08-01  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* examples/Makefile.am:
	* gst/sine/Makefile.am:
	* gst/sine/gstsinesrc.c: (gst_sinesrc_init), (gst_sinesrc_create),
	(gst_sinesrc_set_property), (plugin_init):
	* gst/sine/gstsinesrc.h:
	* gst/volume/Makefile.am:
	* gst/volume/gstvolume.c: (gst_volume_set_volume),
	(gst_volume_set_mute), (gst_volume_dispose), (gst_volume_init),
	(volume_process_float), (volume_process_int16),
	(volume_set_property), (plugin_init):
	* gst/volume/gstvolume.h:
	  deactivate and remove dparams (libgstcontrol)

2005-07-29  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
	Convert me to BaseTransform!! help..

2005-07-29  Andy Wingo  <wingo@pobox.com>

	* ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
	sinks.

	* ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
	support of both endiannesses.

2005-07-28  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
	  Fix confusing debug message (s/event/query/)

2005-07-28  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.h:
	  Use "_stdint.h" instead of <stdint.h>

2005-07-27  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/Makefile.am:
	Revert wrong commit.

2005-07-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
	More compilation fixen.

2005-07-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_event), (gst_base_audio_sink_render),
	(gst_base_audio_sink_create_ringbuffer),
	(gst_base_audio_sink_change_state):
	Fix compilation.

2005-07-27  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (setup_dynamic_link),
	(make_dv_pipeline), (make_vorbis_theora_pipeline), (query_rates),
	(query_positions_elems), (query_positions_pads), (do_seek):
	Update seek example.

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
	(gst_ogg_pad_typefind), (gst_ogg_demux_chain_elem_pad),
	(gst_ogg_demux_queue_data), (gst_ogg_demux_chain_peer),
	(gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
	(gst_ogg_demux_handle_event),
	(gst_ogg_demux_deactivate_current_chain),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
	(gst_ogg_demux_chain), (gst_ogg_demux_send_event),
	(gst_ogg_demux_loop):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
	* ext/theora/theoradec.c: (theora_dec_src_event),
	(theora_dec_src_getcaps), (theora_dec_sink_event),
	(theora_dec_push), (theora_dec_chain):
	* ext/vorbis/Makefile.am:
	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_event),
	(vorbis_dec_sink_event), (vorbis_dec_push),
	(vorbis_handle_data_packet):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_event),
	(gst_vorbisenc_chain):
	* gst/playback/gststreaminfo.c: (cb_probe):
	* gst/subparse/gstsubparse.c: (gst_subparse_src_event):
	* gst/videorate/gstvideorate.c: (gst_videorate_event):
	* gst/videoscale/gstvideoscale.c:
	(gst_videoscale_handle_src_event):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_event):
	* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame),
	(gst_ximagesink_navigation_send_event):
	* sys/xvimage/xvimagesink.c:
	(gst_xvimagesink_navigation_send_event):
	Various event updates and cleanups

2005-07-27  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/videoscale/gstvideoscale.c: (gst_videoscale_prepare_images):
	  Fix segfault for I420/YV12.

2005-07-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
	  Report bitrate.

2005-07-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybin.c: (gen_video_element),
	(gen_audio_element):
	  Switch to auto*sink elements as default sinks; add volume element
	  so that volume control in totem works.

2005-07-21  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (gen_preroll_element):
	* gst/playback/gstplaybin.c: (setup_sinks),
	(gst_play_bin_change_state):
	Refcount fix and more comments.

2005-07-21  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/Makefile.am:
	* sys/ximage/ximage.c: (plugin_init):
	* sys/ximage/ximagesink.c:
	Prepare for adding ximagesrc, rename of plugin to ximage etc.
	

2005-07-21  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
	(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
	(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
	(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
	(gst_ogg_demux_init), (gst_ogg_demux_activate_chain),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info),
	(gst_ogg_demux_collect_info), (gst_ogg_demux_chain),
	(gst_ogg_demux_send_event), (gst_ogg_demux_loop):
	Generate correct disconts for live chained oggs.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render),
	(gst_base_audio_sink_create_ringbuffer),
	(gst_base_audio_sink_change_state):
	Handle discont math correctly.

	* gst/playback/gstplaybin.c: (add_sink):
	Some small debug cleanup.

2005-07-21  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_init), (gst_ogg_pad_event),
	(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
	(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
	(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
	(gst_ogg_demux_init), (gst_ogg_demux_deactivate_current_chain),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_find_pad),
	(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
	(gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
	(gst_ogg_demux_send_event), (gst_ogg_demux_loop),
	(gst_ogg_demux_change_state), (gst_ogg_print):
	Reorganize code to send the right disconts when in streaming
	mode.

2005-07-20  Andy Wingo  <wingo@pobox.com>

	* gst/videoscale/vs_image.c (vs_image_scale_nearest_YUYV): Typo
	fix (?), fixes a seggie mcfalterson (#310894).

2005-07-20  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_headers),
	(gst_ogg_mux_set_header_on_caps):
	* ext/theora/theoraenc.c: (theora_set_header_on_caps):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
	* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_set_channel_positions),
	(gst_audio_set_structure_channel_positions_list):
	* gst/playback/gstdecodebin.c: (dynamic_create):
	* gst/playback/gstplaybasebin.c: (setup_source), (mute_group_type):
	* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
	  Fixes for API changes in core.

2005-07-20  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (fill_buffer):
	  Use _new_custom() so we can set custom message types for buffering
	  messages.

2005-07-20  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/gconf/.cvsignore:
	* gst-libs/gst/gconf/Makefile.am:
	* gst-libs/gst/gconf/test-gconf.c:
	* pkgconfig/Makefile.am:
	* pkgconfig/gstreamer-gconf-uninstalled.pc.in:
	* pkgconfig/gstreamer-gconf.pc.in:
	  Remove gconf stuff, use gconf elements instead from now on.

2005-07-20  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/TODO:
	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	(gst_audio_clock_get_internal_time):
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
	(gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
	(gst_base_audio_sink_render),
	(gst_base_audio_sink_create_ringbuffer),
	(gst_base_audio_sink_change_state):
	Make sure the audio clock always returns an increasing value.

2005-07-19  Andy Wingo  <wingo@pobox.com>

	* gst/videotestsrc/: Cleanups.

2005-07-19  Wim Taymans  <wim@fluendo.com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
	Better debugging.

2005-07-19  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_dv_pipeline),
	(make_vorbis_theora_pipeline), (query_rates),
	(query_positions_elems), (query_positions_pads), (do_seek):
	Make correct DV pipeline.

2005-07-18  Andy Wingo  <wingo@pobox.com>

	* configure.ac (DEFAULT_AUDIOSINK, DEFAULT_AUDIOSRC): Use alsa by
	default. Also because it's the only thing that really works. (This
	is used in the GConf elements).
	Use AS_LIBTOOL_TAGS.

2005-07-18  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (remove_element_chain):
	* gst/playback/gstplaybin.c: (add_sink):
	* gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
	(gst_stream_info_set_mute):
	* gst/playback/gststreamselector.c:
	(gst_stream_selector_get_linked_pad),
	(gst_stream_selector_getcaps), (gst_stream_selector_chain):
	More leak and compile fixes.

2005-07-18  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_vorbis_theora_pipeline),
	(query_rates), (query_positions_elems), (query_positions_pads),
	(do_seek), (seek_cb), (stop_seek):
	Updated seek example. 

	* gst/playback/gstdecodebin.c: (remove_element_chain), (unlinked):
	* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
	(queue_out_of_data), (gen_preroll_element), (new_decoded_pad):
	* gst/playback/gstplaybin.c: (add_sink):
	* gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
	(gst_stream_info_set_mute):
	Some refcount leak fixes.

2005-07-16  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Align samples even if we have roundoff errors in the 
	timestamp conversion.

2005-07-16  Wim Taymans  <wim@fluendo.com>

	* docs/libs/tmpl/gstringbuffer.sgml:
	* examples/seeking/seek.c: (make_vorbis_theora_pipeline),
	(query_rates), (query_positions_elems), (query_positions_pads),
	(update_scale), (do_seek):
	Updated seek example.

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain),
	(gst_ogg_demux_find_chains), (gst_ogg_demux_send_event),
	(gst_ogg_demux_loop):
	Push out correct discont values.

	* ext/theora/theoradec.c: (theora_dec_src_convert),
	(theora_dec_sink_convert), (theora_dec_src_getcaps),
	(theora_dec_sink_event), (theora_handle_type_packet),
	(theora_handle_header_packet), (theora_dec_push),
	(theora_handle_data_packet), (theora_dec_chain),
	(theora_dec_change_state):
	Better timestamping.

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(vorbis_dec_sink_event), (vorbis_dec_push),
	(vorbis_handle_data_packet), (vorbis_dec_chain):
	* ext/vorbis/vorbisdec.h:
	Better timestamping.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times),
	(gst_base_audio_sink_event), (gst_base_audio_sink_render):
	Handle syncing on timestamps instead of sample offsets. Make
	use of DISCONT values as described in design docs.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_get_time):
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire),
	(gst_ring_buffer_set_sample), (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	* gst-libs/gst/audio/gstringbuffer.h:
	* sys/ximage/ximagesink.c: (gst_ximagesink_get_times),
	(gst_ximagesink_show_frame):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
	Correcly convert buffer timestamp to stream time.

2005-07-16  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_get_buffer):
	Timestamp buffers correctly.

	* gst/playback/gstplaybin.c: (gen_video_element):
	Make internal fakesink silent.

2005-07-15  Wim Taymans  <wim@fluendo.com>

	* gst/ffmpegcolorspace/Makefile.am:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_caps_remove_format_info),
	(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
	(gst_ffmpegcsp_get_type), (gst_ffmpegcsp_class_init),
	(gst_ffmpegcsp_init), (gst_ffmpegcsp_get_size),
	(gst_ffmpegcsp_transform_ip), (gst_ffmpegcsp_transform):
	Ported ffmpegcolorspace to basetransform.

	* gst/videoscale/gstvideoscale.c: (gst_videoscale_transform):
	* gst/volume/gstvolume.c: (volume_transform):
	Ported to new API.

2005-07-14  Wim Taymans  <wim@fluendo.com>

	* gst/videotestsrc/Makefile.am:
	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get_type),
	(gst_videotestsrc_class_init), (gst_videotestsrc_negotiate),
	(gst_videotestsrc_setcaps), (gst_videotestsrc_getcaps),
	(gst_videotestsrc_init), (gst_videotestsrc_event),
	(gst_videotestsrc_create), (gst_videotestsrc_start),
	(gst_videotestsrc_stop), (gst_videotestsrc_get_times),
	(gst_videotestsrc_set_pattern), (gst_videotestsrc_set_property),
	(gst_videotestsrc_get_property):
	* gst/videotestsrc/gstvideotestsrc.h:
	Make videotestsrc a pushsrc.

2005-07-14  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstfdset.c: (gst_fdset_free):
	* gst/tcp/gstmultifdsink.c: (gst_multifdsink_init),
	(gst_multifdsink_add), (gst_multifdsink_remove),
	(gst_multifdsink_clear), (gst_multifdsink_get_stats),
	(gst_multifdsink_remove_client_link),
	(gst_multifdsink_client_queue_data),
	(gst_multifdsink_client_queue_caps),
	(gst_multifdsink_client_queue_buffer),
	(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
	(gst_multifdsink_stop):
	* gst/tcp/gstmultifdsink.h:
	0.8 backporting.

	* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
	Also draw image when not from a pool.

2005-07-14  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
	(mute_stream), (silence_stream):
	Small debug additions.

2005-07-14  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose),
	(gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_loop):
	Better error recovery, ignore unconnected pads and
	non-fatal errors.

2005-07-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/libs/tmpl/gstaudio.sgml:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstgconf.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* docs/libs/tmpl/gsttuner.sgml:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get_type),
	(gst_tcpclientsrc_class_init):
	* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get_type),
	(gst_tcpserversrc_class_init):
	* sys/v4l/gstv4lelement.c:
	  more autistic cleanliness in functions/names/defines

2005-07-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
	  added manually to each Makefile.am so we are sure it goes
	  *last* and doesn't add -L flags before linking in libs of our
	  own, like, say, internal .la libs, that then accidentally pick
	  up the installed copy.
	* docs/libs/Makefile.am:
	* ext/alsa/Makefile.am:
	* ext/cdparanoia/Makefile.am:
	* ext/gnomevfs/Makefile.am:
	* ext/libvisual/Makefile.am:
	* ext/ogg/Makefile.am:
	* ext/theora/Makefile.am:
	* ext/vorbis/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	* gst/adder/Makefile.am:
	* gst/audioconvert/Makefile.am:
	* gst/audiorate/Makefile.am:
	* gst/audioscale/Makefile.am:
	* gst/ffmpegcolorspace/Makefile.am:
	* gst/playback/Makefile.am:
	* gst/sine/Makefile.am:
	* gst/subparse/Makefile.am:
	* gst/tags/Makefile.am:
	* gst/tcp/Makefile.am:
	* gst/typefind/Makefile.am:
	* gst/videorate/Makefile.am:
	* gst/videoscale/Makefile.am:
	* gst/videotestsrc/Makefile.am:
	* gst/volume/Makefile.am:
	* sys/v4l/Makefile.am:
	* sys/ximage/Makefile.am:
	* sys/xvimage/Makefile.am:
	  adapt properly to this change. This should make sure that
	  plugins and libs properly link to the as-yet-uninstalled
	  copies of stuff like libgstinterfaces and libgstvideo

2005-07-13  Andy Wingo  <wingo@pobox.com>

	* sys/v4l/gstv4lsrc.c (gst_v4lsrc_stop): Fix a spurious warning.
	(gst_v4lsrc_fixate): Fixate on format as well.

	* sys/xvimage/xvimagesink.c (gst_xvimage_buffer_destroy) 
	(gst_xvimagesink_xvimage_new): Ref the xvimagesink while the
	buffer points to it.
	(gst_xvimagesink_check_xshm_calls): Don't use our xvimage buffer,
	rather just doing X calls ourselves. Also fixes a memleak.

2005-07-12  Andy Wingo  <wingo@pobox.com>

	* sys/v4l/gstv4lsrc.c (gst_v4lsrc_get_property) 
	(gst_v4lsrc_set_property, gst_v4lsrc_class_init, gst_v4lsrc_init) 
	(gst_v4lsrc_create): Re-add the copy-mode property, default to
	TRUE to avoid deadlocks if an element holds on to our buffers.

2005-07-11  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
	(gst_sinesrc_init), (gst_sinesrc_create),
	(gst_sinesrc_set_property), (gst_sinesrc_get_property),
	(gst_sinesrc_start):
	* gst/sine/gstsinesrc.h:
	  removing num-buffers property before moving it

2005-07-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  use overridable ERROR_CFLAGS
	* docs/libs/gst-plugins-base-libs.types:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
	(gst_alsasink_class_init):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_type),
	(gst_alsasrc_class_init):
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
	(gst_audio_filter_base_init), (gst_audio_filter_class_init),
	(gst_audio_filter_link), (gst_audio_filter_init),
	(gst_audio_filter_chain), (gst_audio_filter_set_property),
	(gst_audio_filter_get_property),
	(gst_audio_filter_class_add_pad_templates):
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	(gst_audio_filter_template_get_type),
	(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), (plugin_init),
	(gst_audio_filter_template_setup),
	(gst_audio_filter_template_filter),
	(gst_audio_filter_template_filter_inplace):
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_acquire),
	(gst_audioringbuffer_release), (gst_audioringbuffer_start),
	(gst_audioringbuffer_stop), (gst_audioringbuffer_delay),
	(gst_audio_sink_base_init), (gst_audio_sink_class_init),
	(gst_audio_sink_init), (gst_audio_sink_create_ringbuffer):
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_get_type),
	(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_start), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audio_src_base_init),
	(gst_audio_src_class_init), (gst_audio_src_init),
	(gst_audio_src_create_ringbuffer):
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_base_init), (gst_base_audio_sink_class_init),
	(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
	(gst_base_audio_sink_get_clock), (gst_base_audio_sink_get_time),
	(gst_base_audio_sink_set_property),
	(gst_base_audio_sink_get_property), (gst_base_audio_sink_setcaps),
	(gst_base_audio_sink_get_times), (gst_base_audio_sink_event),
	(gst_base_audio_sink_preroll), (gst_base_audio_sink_render),
	(gst_base_audio_sink_create_ringbuffer),
	(gst_base_audio_sink_callback), (gst_base_audio_sink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_base_init), (gst_base_audio_src_class_init),
	(gst_base_audio_src_init), (gst_base_audio_src_get_clock),
	(gst_base_audio_src_get_time), (gst_base_audio_src_set_property),
	(gst_base_audio_src_get_property), (gst_base_audio_src_fixate),
	(gst_base_audio_src_setcaps), (gst_base_audio_src_get_times),
	(gst_base_audio_src_event), (gst_base_audio_src_create),
	(gst_base_audio_src_create_ringbuffer),
	(gst_base_audio_src_callback), (gst_base_audio_src_change_state):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
	(gst_ring_buffer_class_init), (gst_ring_buffer_init),
	(gst_ring_buffer_dispose), (gst_ring_buffer_finalize),
	(gst_ring_buffer_debug_spec_caps),
	(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
	(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
	(gst_ring_buffer_start), (gst_ring_buffer_pause),
	(gst_ring_buffer_stop), (gst_ring_buffer_delay),
	(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
	(wait_segment), (gst_ring_buffer_commit), (gst_ring_buffer_read),
	(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
	(gst_ring_buffer_clear):
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
	(gst_video_sink_class_init), (gst_video_sink_get_type):
	* gst-libs/gst/video/videosink.h:
	* gst/tcp/gstmultifdsink.c: (gst_multifdsink_get_type),
	(gst_multifdsink_class_init),
	(gst_multifdsink_handle_client_write),
	(gst_multifdsink_change_state):
	* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
	(gst_tcpclientsink_setcaps):
	* sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
	(gst_ximagesink_getcaps), (gst_ximagesink_setcaps),
	(gst_ximagesink_change_state), (gst_ximagesink_show_frame),
	(gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc),
	(gst_ximagesink_send_pending_navigation),
	(gst_ximagesink_set_xwindow_id), (gst_ximagesink_get_desired_size),
	(gst_ximagesink_class_init), (gst_ximagesink_get_type):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps),
	(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_send_pending_navigation),
	(gst_xvimagesink_navigation_send_event),
	(gst_xvimagesink_set_xwindow_id),
	(gst_xvimagesink_get_desired_size), (gst_xvimagesink_class_init),
	(gst_xvimagesink_get_type):
	more macro splitting

2005-07-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
	  plug a memleak, allows me to import 1479 albums in one go
	  in jamboree
	* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	(vorbis_handle_type_packet), (vorbis_dec_chain),
	(vorbis_dec_change_state):
	  fix some format strings

2005-07-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	(gst_alsasink_set_property), (gst_alsasink_get_property):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
	(gst_alsasrc_set_property), (gst_alsasrc_get_property):
	  add device property

2005-07-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
	* ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
	(audiocast_register_listener), (audiocast_thread_run),
	(gst_gnomevfssrc_send_additional_headers_callback),
	(gst_gnomevfssrc_received_headers_callback),
	(gst_gnomevfssrc_push_callbacks), (gst_gnomevfssrc_pop_callbacks),
	(gst_gnomevfssrc_get_icy_metadata), (gst_gnomevfssrc_create),
	(gst_gnomevfssrc_get_size):
	  add/clean up debugging
	* gst/audiorate/gstaudiorate.c: (gst_audiorate_init):
	  cleanups

2005-07-07  Andy Wingo  <wingo@pobox.com>

	* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Also fixate the
	framerate. Need to get a handle on when exactly this function is
	called, tho.

	* sys/v4l/v4lsrc_calls.h:
	* sys/v4l/v4lsrc_calls.c: Remove sync-related stuff.
	(gst_v4lsrc_get_fps_list): Moved here from gstv4lsrc.c.
	(gst_v4lsrc_buffer_new): Totally derive from GstBuffer.

	* sys/v4l/v4l_calls.h: Cast to V4lElement.
	* sys/v4l/v4l_calls.c: Header loc fixen, don't load mjpeg, all
	v4lelements are sources.

	* sys/v4l/gstv4lxoverlay.h:
	* sys/v4l/gstv4lxoverlay.c:
	* sys/v4l/gstv4ltuner.h:
	* sys/v4l/gstv4ltuner.c: Header loc fixen.
	
	* sys/v4l/gstv4lsrc.h:
	* sys/v4l/gstv4lsrc.c: Crucial GPL update. Clean up a bit, port to
	PushSrc/BaseSrc. Removed most sync-related properties, videorate
	or something should handle that. Made a live source.

	* sys/v4l/gstv4lelement.h:
	* sys/v4l/gstv4lelement.c: Derive from GstPushSrc. No more
	signals. Some cleanups.

	* sys/v4l/gstv4lcolorbalance.h: Interface header update.

	* sys/v4l/gstv4l.c: Don't register v4lelement, or the jpeg/mjpeg
	stuff.

	* sys/v4l/Makefile.am: Build everything except the jpeg/mjpeg
	stuff.

	* sys/Makefile.am (SUBDIRS): Hit the V4L crack pipe.

2005-07-07  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (theora_get_query_types),
	(theora_dec_src_getcaps), (theora_dec_push):
	* ext/vorbis/vorbisdec.c: (vorbis_get_query_types):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types):
	Remove deprecated/unused query types.

2005-07-06  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/Makefile.am:
	* ext/alsa/gstalsaplugin.c: (plugin_init):
	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_type),
	(gst_alsasrc_dispose), (gst_alsasrc_base_init),
	(gst_alsasrc_class_init), (gst_alsasrc_init),
	(gst_alsasrc_getcaps), (set_hwparams), (set_swparams),
	(alsasrc_parse_spec), (gst_alsasrc_open), (gst_alsasrc_close),
	(xrun_recovery), (gst_alsasrc_read), (gst_alsasrc_delay),
	(gst_alsasrc_reset):
	* ext/alsa/gstalsasrc.h:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(gst_audioringbuffer_start):
	* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_get_type),
	(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
	(gst_audioringbuffer_init), (gst_audioringbuffer_dispose),
	(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
	(gst_audioringbuffer_release), (gst_audioringbuffer_start),
	(gst_audioringbuffer_stop), (gst_audioringbuffer_delay),
	(gst_audiosrc_base_init), (gst_audiosrc_class_init),
	(gst_audiosrc_init), (gst_audiosrc_create_ringbuffer):
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
	(gst_baseaudiosink_get_time), (gst_baseaudiosink_setcaps),
	(gst_baseaudiosink_preroll), (gst_baseaudiosink_render):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_baseaudiosrc_base_init), (gst_baseaudiosrc_class_init),
	(gst_baseaudiosrc_init), (gst_baseaudiosrc_get_clock),
	(gst_baseaudiosrc_get_time), (gst_baseaudiosrc_set_property),
	(gst_baseaudiosrc_get_property), (gst_baseaudiosrc_fixate),
	(gst_baseaudiosrc_setcaps), (gst_baseaudiosrc_get_times),
	(gst_baseaudiosrc_event), (gst_baseaudiosrc_create),
	(gst_baseaudiosrc_create_ringbuffer), (gst_baseaudiosrc_callback),
	(gst_baseaudiosrc_change_state):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ringbuffer_debug_spec_caps), (gst_ringbuffer_debug_spec_buff),
	(gst_ringbuffer_parse_caps), (gst_ringbuffer_start),
	(gst_ringbuffer_pause), (gst_ringbuffer_stop),
	(gst_ringbuffer_samples_done), (gst_ringbuffer_set_sample),
	(wait_segment), (gst_ringbuffer_commit), (gst_ringbuffer_read),
	(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance):
	* gst-libs/gst/audio/gstringbuffer.h:
	Added audiosource base classes.
	Ported alsasrc, still very basic.

2005-07-06  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (theora_dec_src_getcaps),
	(theora_dec_push), (theora_handle_data_packet):
	Prepare for better timestamp fix later.

	* gst/audioconvert/gstaudioconvert.c:
	List most accurate caps first

	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
	Use proper pad task function.

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_show_frame):
	Fix deadlock when alloc failed.

2005-07-05  Andy Wingo  <wingo@pobox.com>

	* ext/gnomevfs/gstgnomevfssrc.c:
	* gst/sine/gstsinesrc.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/tcp/gsttcpclientsrc.c: s/BASESRC/BASE_SRC/.

	* sys/v4l/: Port from 0.8.

	* Many files: Null if we got it....

2005-07-05  Andy Wingo  <wingo@pobox.com>

	* gst/tcp/gsttcpserversink.c (gst_tcpserversink_handle_server_read): 
	* gst/tcp/gstmultifdsink.c (gst_multifdsink_client_queue_data):
	Signedness fixes.

2005-07-05  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	* gst/tcp/Makefile.am:
	* gst/tcp/README:
	* gst/tcp/gstmultifdsink.c: (gst_multifdsink_get_type),
	(gst_multifdsink_base_init), (gst_multifdsink_class_init),
	(gst_multifdsink_init), (gst_multifdsink_remove_client_link),
	(is_sync_frame), (gst_multifdsink_handle_client_write),
	(gst_multifdsink_render), (gst_multifdsink_start),
	(gst_multifdsink_stop), (gst_multifdsink_change_state):
	* gst/tcp/gstmultifdsink.h:
	* gst/tcp/gsttcp.c: (gst_tcp_host_to_ip),
	(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps),
	(gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
	* gst/tcp/gsttcp.h:
	* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
	(gst_tcpclientsink_init), (gst_tcpclientsink_setcaps),
	(gst_tcpclientsink_render), (gst_tcpclientsink_start),
	(gst_tcpclientsink_stop), (gst_tcpclientsink_change_state):
	* gst/tcp/gsttcpclientsink.h:
	* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get_type),
	(gst_tcpclientsrc_base_init), (gst_tcpclientsrc_class_init),
	(gst_tcpclientsrc_init), (gst_tcpclientsrc_getcaps),
	(gst_tcpclientsrc_create), (gst_tcpclientsrc_start),
	(gst_tcpclientsrc_stop), (gst_tcpclientsrc_unlock):
	* gst/tcp/gsttcpclientsrc.h:
	* gst/tcp/gsttcpplugin.c: (plugin_init):
	* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init):
	* gst/tcp/gsttcpserversink.h:
	* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get_type),
	(gst_tcpserversrc_base_init), (gst_tcpserversrc_class_init),
	(gst_tcpserversrc_init), (gst_tcpserversrc_finalize),
	(gst_tcpserversrc_create), (gst_tcpserversrc_start),
	(gst_tcpserversrc_stop):
	* gst/tcp/gsttcpserversrc.h:
	* gst/tcp/gsttcpsink.c:
	* gst/tcp/gsttcpsink.h:
	* gst/tcp/gsttcpsrc.c:
	* gst/tcp/gsttcpsrc.h:
	Ported tcp plugins to 0.9. 
	

2005-07-05  Andy Wingo  <wingo@pobox.com>

	* gst/playback/gstplaybasebin.c (fill_buffer):
	message_new_application fixen.

	* gst/videoscale/gstvideoscale.c (gst_videoscale_transform_caps):
	Style fix.

2005-07-04  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
	Set caps on output buffer.

2005-07-04  Andy Wingo  <wingo@pobox.com>

	* ext/gnomevfs/gstgnomevfssrc.c
	(gst_gnomevfssrc_received_headers_callback) 
	(audiocast_thread_kill, audiocast_thread_run): FORTIFY fixen,
	hopefully.

	* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
	No refcount leakage.

	* configure.ac: Enable -Werror.
	
	* ext/theora/theoradec.c (theora_dec_src_getcaps):
	* gst/audioconvert/bufferframesconvert.c
	(buffer_frames_convert_fixate):
	* gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int)
	(gst_audio_convert_fixate):
	* gst/sine/gstsinesrc.c (gst_sinesrc_src_fixate)
	(gst_sinesrc_create): Fixate func changes.
	
	* sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
	(gst_ximagesink_buffer_alloc): Unused var.

2005-07-01  Andy Wingo  <wingo@pobox.com>

	* ext/theora/theoradec.c (theora_dec_src_getcaps): Implement a
	getcaps to do explicit caps. Needs to be done in all decoders,
	possibly via a base class.

	* configure.ac (GST_PLUGIN_LDFLAGS): Add videoscale.

	* ext/ogg/gstoggdemux.c (gst_ogg_pad_typefind): No need to set
	caps on the sink pad, just rely on the pad template. Also, setting
	ANY caps on a pad is not valid because the caps are not fixed.

	* sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc): Set the
	caps on the buffer, and get the width from the desired_caps if
	they're set.
	(gst_ximagesink_renegotiate_size): Implement via setting the
	desired_caps on the ximagesink.
	(gst_ximagesink_setcaps): Only reset the width of the player if it
	wasn't already set. Not sure if this is right.
	(gst_ximagesink_show_frame): Memcpy only for normal buffers.

	* sys/ximage/ximagesink.h (desired_caps): New field, is the caps
	that the user wants. NULL unless the window has been resized.

	* gst/volume/gstvolume.c (volume_transform): Adapt to
	basetransform refcount changes.
	
2005-07-01  Andy Wingo  <wingo@pobox.com>

	* gst/videoscale/gstvideoscale.c:
	* gst/videoscale/gstvideoscale.h: Clean up, port to 0.9. Derives
	from BaseTransform, implements a transform_caps. Removed dead code
	including some PAR stuff that was never reached -- should probably
	be added back somehow.

2005-07-01  Andy Wingo  <wingo@pobox.com>

	* gst/videoscale: Merge David's work from 0.8 branch. Changes to
	come later.

2005-06-30  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-libs.types:
	* ext/alsa/Makefile.am:
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixeroptions.h:
	* ext/alsa/gstalsamixertrack.h:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/colorbalance/.cvsignore:
	* gst-libs/gst/colorbalance/Makefile.am:
	* gst-libs/gst/colorbalance/colorbalance-marshal.list:
	* gst-libs/gst/colorbalance/colorbalance.c:
	* gst-libs/gst/colorbalance/colorbalance.h:
	* gst-libs/gst/colorbalance/colorbalance.vcproj:
	* gst-libs/gst/colorbalance/colorbalancechannel.c:
	* gst-libs/gst/colorbalance/colorbalancechannel.h:
	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/colorbalance.c:
	(gst_color_balance_class_init):
	* gst-libs/gst/interfaces/colorbalance.h:
	* gst-libs/gst/interfaces/interfaces-marshal.list:
	* gst-libs/gst/interfaces/mixer.c: (gst_mixer_class_init):
	* gst-libs/gst/interfaces/mixer.h:
	* gst-libs/gst/interfaces/mixeroptions.h:
	* gst-libs/gst/interfaces/navigation.c:
	* gst-libs/gst/interfaces/tuner.c: (gst_tuner_class_init):
	* gst-libs/gst/interfaces/tuner.h:
	* gst/volume/Makefile.am:
	* gst/volume/gstvolume.c:
	* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
	* sys/ximage/Makefile.am:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/Makefile.am:
	* sys/xvimage/xvimagesink.c:
	  fold in all interfaces into an interfaces dir, preserving CVS
	  history

2005-06-30  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Fix build after riff changes.

2005-06-30  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	(gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
	(gst_riff_create_video_template_caps),
	(gst_riff_create_audio_template_caps),
	(gst_riff_create_iavs_template_caps):
	* gst-libs/gst/riff/riff-media.h:
	* gst-libs/gst/riff/riff-read.h:
	* gst-libs/gst/riff/riff.c: (gst_riff_init):
	  Add gst_riff_init() to initialize the debug category, instead
	  of plugin_init(). Port riff-media.[ch] from -THREADED to HEAD.

2005-06-29  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init):
	  Oops, I shouldn't apply hacks.

2005-06-29  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init):
	  Remove pad_loop function which doesn't work.

2005-06-29  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain):
	  Send EOS when deactivating.
	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	(check_queue), (queue_threshold_reached), (queue_out_of_data),
	(gen_preroll_element), (probe_triggered), (mute_stream),
	(silence_stream), (new_decoded_pad), (setup_substreams),
	(set_active_source):
	* gst/playback/gstplaybin.c: (gst_play_bin_get_property),
	(remove_sinks), (add_sink):
	* gst/playback/gststreaminfo.c: (cb_probe), (gst_stream_info_new):
	  Change for new probe API.

2005-06-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
	(gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ringbuffer_set_callback):
	Fix compilation error.
	Ringbuffer starts out as not running.
	Free our clock in dispose.
	When releasing the ringbuffer we need to renegotiate so
	clear the pad caps.

2005-06-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* autogen.sh:
	* configure.ac:
	* docs/Makefile.am:
	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-libs-docs.sgml:
	* docs/libs/gst-plugins-libs-sections.txt:
	* docs/libs/gst-plugins-libs.types:
	* docs/libs/tmpl/gstaudio.sgml:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ringbuffer_set_callback):
	  reinstate gtk-doc docs for plugin libs

2005-06-28  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_init):
	Removed pad loop function.

2005-06-28  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
	If we're building a chain we are not in an error case
	when we queue a buffer.

2005-06-28  Andy Wingo  <wingo@pobox.com>

	* *.c: Don't cast to GstObject before reffing/unreffing.

2005-06-27  Andy Wingo  <wingo@pobox.com>

	* gst/videotestsrc/gstvideotestsrc.c
	(gst_videotestsrc_activate_push): Activation API changes.

	* gst/playback/gstdecodebin.c (gst_decode_bin_change_state) 
	(gst_decode_bin_dispose): Free dynamics in READY->NULL, because
	they have refs on the decodebin.

	* ext/ogg/gstoggdemux.c (gst_ogg_pad_class_init): Ref the right
	parent class.
	(gst_ogg_pad_typefind): Don't leak a pad ref.
	(gst_ogg_chain_new_stream): gst_object_unref, not g_object_unref.
	(gst_ogg_demux_sink_activate, gst_ogg_demux_sink_activate_push) 
	(gst_ogg_demux_sink_activate_pull): Changes for activation API.

2005-06-27  Edward Hervey  <edward@fluendo.com>

	* ext/theora/theoradec.c: (theora_dec_change_state): 
	re-arranged call to parent's state change in order to avoid locks (or
	worse).

2005-06-26  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
	2nd argument of 'unknow-type' signal is a GstCaps and not a
	GstMiniObject

2005-06-25  Jan Schmidt  <thaytan@mad.scientist.com>
	* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
	  Set the worker thread's running flag to TRUE before starting the
	  thread.
	* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
	  Catch a failure to add typefind to the bin.

2005-06-24  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
	(gst_sinesrc_init), (gst_sinesrc_create),
	(gst_sinesrc_set_property), (gst_sinesrc_get_property),
	(gst_sinesrc_start):
	* gst/sine/gstsinesrc.h:
	  add num-buffers and timestamp-offset properties
	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_videotestsrc_class_init), (gst_videotestsrc_set_property),
	(gst_videotestsrc_get_property):
	  add timestamp-offset property

2005-06-23  Christian Schaller  <uraeus@gnome.org>

	* configure.ac: add videorate
	* gst-plugins-base.spec.in: add videorate

2005-06-23  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
	(gst_videorate_getcaps), (gst_videorate_setcaps),
	(gst_videorate_event), (gst_videorate_chain):
	Fixed videorate, fixating an already fixated caps is not
	an error.

2005-06-23  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/README:
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_set_header_on_caps):
	Buffer on caps is not boxed anymore.

2005-06-22  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoraenc.c: (theora_set_header_on_caps):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
	Set buffers on caps as miniobjects and not as boxed.

2005-06-19  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  back to HEAD

=== release 0.9.1 ===

2005-06-19  Thomas Vander Stichele  <thomas at apestaart dot org>

	* .cvsignore:
	* NEWS:
	* README:
	* RELEASE:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  updates for release

2005-06-09  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
	
2005-06-09  Andy Wingo  <wingo@pobox.com>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/net/Makefile.am:
	Add gstnet to build.

2005-06-09  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/gconf/gconf.c:
	* gst/playback/test.c:
	* gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
	fixes.

	* gst/audioconvert/gstaudioconvert.c: RPAD fixes.

	* ext/theora/theoraenc.c (theora_enc_chain): 
	* ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.

	* ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
	RealPad.

2005-06-02  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/net/Makefile.am:
	* pkgconfig/gstreamer-libs-uninstalled.pc.in:
	* pkgconfig/gstreamer-libs.pc.in:
	Added net stuff, version net lib.

2005-06-02  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_vorbis_theora_pipeline),
	(query_rates), (query_positions_elems), (query_positions_pads),
	(do_seek):
	Updated seek example.

2005-06-02  Andy Wingo  <wingo@pobox.com>

	* pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
	* pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
	list.

	* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Don't
	remove the typefind, the bin dispose will do it for us. When it's
	removed and unreffed, the signal handler will be disconnected,
	too.
	(unlinked): It's too difficult to disconnect from unlinked
	handlers, as they are on pads not elements. Just punt if the pads
	aren't grandkids of the bin.

2005-06-02  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/README:
	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_clear_chains):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
	* ext/theora/theoradec.c: (theora_dec_src_query),
	(theora_handle_data_packet):
	* ext/theora/theoraenc.c: (theora_buffer_from_packet),
	(theora_enc_chain):
	* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	(vorbis_handle_data_packet):
	* gst/audioconvert/bufferframesconvert.c:
	(buffer_frames_convert_chain):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
	(gst_ffmpegcsp_chain):
	* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
	(gst_videorate_getcaps), (gst_videorate_setcaps),
	(gst_videorate_event), (gst_videorate_chain):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_activate),
	(gst_videotestsrc_src_query), (gst_videotestsrc_loop):
	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
	(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
	(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
	Cleanups and buffer alloc.

2005-05-31  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_delay):
	Don't try to call the delay method when the device is not
	opened.

2005-05-31  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_open):
	Get actual segment size and buffer size after opening
	the device.

2005-05-30  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_clear_chains):
	Also FLUSH upstream, makes the loop function exit faster.
	
	* ext/theora/theoradec.c: (theora_dec_src_query):
	Some more debug info in the query.
	
	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_setcaps):
	Release lock on par error, better error reporting.

2005-05-26  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_chain),
	(gst_ogg_demux_clear_chains), (gst_ogg_demux_change_state):
	Clear chains in READY
	Queue packets until the chain is activated.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
	(gst_audiosink_create_ringbuffer):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
	(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
	(gst_baseaudiosink_set_property), (build_linear_format),
	(debug_spec_caps), (debug_spec_buffer),
	(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
	(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
	(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
	(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
	(gst_ringbuffer_play), (gst_ringbuffer_pause),
	(gst_ringbuffer_stop), (gst_ringbuffer_delay),
	(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
	(wait_segment), (gst_ringbuffer_commit),
	(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
	(gst_ringbuffer_clear):
	Various small cleanups.

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_change_state):
	* gst/subparse/gstsubparse.c: (gst_subparse_chain):
	No need to take the locks anymore.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
	(gst_decode_bin_dispose), (try_to_link_1), (get_our_ghost_pad),
	(remove_element_chain), (no_more_pads), (unlinked), (close_link),
	(type_found):
	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_dispose),
	(group_destroy), (group_commit), (queue_overrun),
	(gen_preroll_element), (no_more_pads), (preroll_unlinked),
	(mute_stream), (new_decoded_pad), (setup_substreams),
	(setup_source), (mute_group_type), (set_active_source),
	(gst_play_base_bin_change_state):
	* gst/playback/gstplaybin.c: (gst_play_bin_dispose),
	(gen_video_element), (gen_text_element), (gen_audio_element),
	(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks):
	* gst/playback/gststreaminfo.c: (gst_stream_info_new),
	(gst_stream_info_dispose), (gst_stream_info_set_mute):
	* gst/playback/gststreamselector.c: (gst_stream_selector_chain):
	Some playbin cleanups mostly refcounting sloppyness.

2005-05-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
	  Work with streaming input.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
	(gst_ffmpegcsp_chain), (gst_ffmpegcsp_change_state):
	No need to take the STREAM lock anymore.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose),
	(gst_ogg_pad_typefind), (gst_ogg_pad_submit_packet),
	(gst_ogg_chain_new_stream), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_chain), (gst_ogg_demux_loop),
	(gst_ogg_demux_sink_activate):
	* ext/theora/theoradec.c: (theora_dec_src_event),
	(theora_handle_comment_packet), (theora_dec_chain),
	(theora_dec_change_state):
	* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	(vorbis_handle_data_packet), (vorbis_dec_chain),
	(vorbis_dec_change_state):
	Remove STREAM locks as they are taken in core now.
	Never set bogus granulepos on vorbis/theora.
	Fix leaks in theoradec tag parsing.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
	Fix memleaks, GST_BUFFER_DATA() is not freed.

2005-05-25  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	Open non-blocking, set to blocking mode afterwards to avoid
	lockups when audio device is busy.

2005-05-23  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
	  This can't be good.

2005-05-23  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
	(gst_audio_convert_chain), (gst_audio_convert_link_src),
	(gst_audio_convert_setcaps):
	  Implement instant setup switching.

2005-05-19  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (probe_triggered):
	  Fix missing unlock.
	* gst/playback/gstplaybin.c: (add_sink):
	  First add, then link (otherwise pad link fails).

2005-05-19  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* examples/Makefile.am:
	fix buildbot (make distcheck)

2005-05-19  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybin.c: (gen_vis_element):
	  Remove some wrong code. Doesn't work yet.

2005-05-19  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/net/Makefile.am:
	* gst-libs/gst/net/README:
	* gst-libs/gst/net/gstnetbuffer.c: (gst_netbuffer_get_type),
	(gst_netbuffer_class_init), (gst_netbuffer_init),
	(gst_netbuffer_finalize), (gst_netbuffer_copy),
	(gst_netbuffer_new), (gst_netaddress_set_ip4_address),
	(gst_netaddress_set_ip6_address), (gst_netaddress_get_net_type),
	(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address):
	* gst-libs/gst/net/gstnetbuffer.h:
	Added buffer subclass to store extra to/from addresses for
	network sources/sinks.

2005-05-18  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
	  Don't lock an unassigned variable.

2005-05-18  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (gen_preroll_element):
	  Increase buffer for video, decrease buffer for other media types.
	* gst/playback/gstplaybin.c: (gen_video_element),
	(gen_audio_element):
	  Change names for debugging purposes.

2005-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
	(gst_ffmpegcsp_chain):
	Enable buffer alloc passthrough if the source and dest
	formats are the same.

2005-05-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
	(gst_ogg_demux_chain_unlocked):
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
	(gst_audio_convert_fixate), (gst_audio_convert_change_state):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_init),
	(gst_ffmpegcsp_bufferalloc), (gst_ffmpegcsp_chain),
	(gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
	(gst_ffmpegcsp_get_property):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
	(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
	(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_xvimage_put), (gst_xvimagesink_imagepool_clear),
	(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_free),
	(gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id):
	Leak fixes in oggdemux.
	Some cleanups in audioconvert.
	Make passthrough work along with buffer_alloc etc.
	Make buffer_alloc and buffer recycling actually work in
	xvimagesink.

2005-05-17  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/subparse/gstsubparse.c: (parse_subrip), (parse_mpsub):
	  make the compiler happy

2005-05-17  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
	(gst_xvimage_buffer_init), (gst_xvimage_buffer_class_init),
	(gst_xvimage_buffer_get_type), (gst_xvimagesink_check_xshm_calls),
	(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_imagepool_clear), (gst_xvimagesink_setcaps),
	(gst_xvimagesink_change_state), (gst_xvimagesink_show_frame),
	(gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_set_xwindow_id):
	* sys/xvimage/xvimagesink.h:
	Port xvimagesink to new MiniObject.

2005-05-17  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
	(gst_audiofilter_chain):
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
	(gst_audiosink_create_ringbuffer):
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
	(gst_audio_convert_fixate), (gst_audio_convert_channels):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
	Fix passthrough in ffmpegcolorspace.
	Fix memset in audiosink on wrong memory.

2005-05-16  David Schleef  <ds@schleef.org>

	* gst/playback/gststreaminfo.c: (cb_probe): Port from GstData
	to GstMiniObject.

2005-05-16  David Schleef  <ds@schleef.org>

	Port from GstData to GstMiniObject.
	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
	(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
	(gst_ogg_mux_collected):
	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	* ext/theora/theoradec.c: (theora_handle_comment_packet),
	(theora_handle_data_packet):
	* ext/theora/theoraenc.c: (theora_buffer_from_packet),
	(theora_set_header_on_caps), (theora_enc_chain):
	* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	(vorbis_handle_comment_packet):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
	* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_get_buffer):
	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
	* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
	(mute_stream), (silence_stream):
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	* gst/volume/gstvolume.c: (volume_transform):
	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
	(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
	(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
	(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
	(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
	(gst_ximagesink_buffer_alloc):
	* sys/ximage/ximagesink.h:

2005-05-12  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(fill_buffer), (check_queue), (queue_threshold_reached),
	(queue_out_of_data):
	* gst/playback/gstplaybasebin.h:
	  Post buffer-fullness on the bus.

2005-05-12  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	(try_to_link_1):
	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(group_commit), (probe_triggered), (setup_source),
	(gst_play_base_bin_change_state):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_init), (remove_sinks), (setup_sinks),
	(gst_play_bin_change_state):
	  Move setup_output_pads into a virtual function, remove
	  group-switch (no longer needed) and redirect (handled by bus
	  now) signals.

2005-05-12  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
	(gst_play_base_bin_class_init), (gst_play_base_bin_finalize),
	(get_active_group), (get_building_group), (group_destroy),
	(group_commit), (check_queue), (queue_overrun),
	(queue_threshold_reached), (queue_out_of_data),
	(gen_preroll_element), (remove_groups), (unknown_type),
	(add_element_stream), (no_more_pads), (probe_triggered),
	(preroll_unlinked), (new_decoded_pad), (setup_subtitle),
	(setup_substreams), (setup_source), (finish_source),
	(prepare_output), (muted_group_change_state),
	(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
	(gst_play_base_bin_change_state):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_init), (gst_play_bin_set_property),
	(gen_video_element), (gen_text_element), (gen_audio_element),
	(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
	(gst_play_bin_change_state):
	* gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
	(cb_probe), (gst_stream_info_new), (gst_stream_info_dispose),
	(stream_info_change_state), (gst_stream_info_set_mute),
	(gst_stream_info_get_property):
	* gst/playback/gststreaminfo.h:
	* gst/playback/gststreamselector.c: (gst_stream_selector_init),
	(gst_stream_selector_get_linked_pad),
	(gst_stream_selector_getcaps),
	(gst_stream_selector_get_linked_pads),
	(gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
	* gst/playback/gststreamselector.h:
	  Rough port of playbin. Needs some more work, but is mostly done,
	  and uses a few locks in important places, which should make stuff
	  like chain-switches clean. Still uses GST_STATE() in a few places,
	  which isn't all that good an idea, subtitles/elements disabled
	  because no elements to test with and thus probably broken, query
	  and event handling moved to GstBin, internal thread removed
	  alltogether because the pipeline does that for us now. Can play
	  Ogg/Vorbis files. Haven't tested anything else yet.

2005-05-12  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
	  Do no-more-pads (needed for autoplugging).

2005-05-10  Andy Wingo  <wingo@pobox.com>

	* ext/vorbis/vorbisdec.c (vorbis_handle_comment_packet): Post a
	message to the bus with the tags. Still not sent downstream tho.

	* gst/playback/gstdecodebin.c (remove_element_chain): Unref after
	get_parent.
	(remove_element_chain): Use OBJECT_PARENT instead of get_parent to
	avoid refcounting hassles.

2005-05-09  Andy Wingo  <wingo@pobox.com>

	* gst/volume/Makefile.am:
	* gst/volume/demo.c
	* gst/volume/gstvolume.h
	* gst/volume/gstvolume.c: Port to 0.9 API, derive from
	basetransform. Probably need an audio filter base class.

2005-05-09  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
	(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
	(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
	(gst_vorbisenc_chain):
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
	(gst_audio_convert_fixate), (gst_audio_convert_channels):
	Make caps writable before writing to it.
	Fix negotiation in audioconvert some more.

2005-05-09  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
	(gst_videorate_getcaps), (gst_videorate_setcaps),
	(gst_videorate_event), (gst_videorate_chain):
	Better negotiation.

2005-05-09  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
	(gst_videorate_getcaps), (gst_videorate_setcaps),
	(gst_videorate_blank_data), (gst_videorate_init),
	(gst_videorate_event), (gst_videorate_chain),
	(gst_videorate_change_state):
	Port videorate, do a better job at negotiation while we're at
	it.

2005-05-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	  Disable libvisual

	* examples/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	  Fixups for missing variables.

2005-05-09  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_theora_pipeline),
	(make_vorbis_theora_pipeline), (make_avi_msmpeg4v3_mp3_pipeline),
	(query_rates), (query_positions_elems), (query_positions_pads),
	(update_scale), (play_cb), (pause_cb), (stop_cb), (main):
	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
	(gst_ogg_pad_query_types), (gst_ogg_pad_src_query),
	(gst_ogg_pad_typefind), (gst_ogg_demux_init),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
	(gst_ogg_demux_read_end_chain), (gst_ogg_demux_sink_activate):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
	(gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
	(gst_ogg_mux_push_page), (gst_ogg_mux_queue_pads),
	(gst_ogg_mux_get_headers), (gst_ogg_mux_send_headers),
	(gst_ogg_mux_collected), (gst_ogg_mux_change_state):
	* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
	(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	(gst_ogm_parse_sink_query), (gst_ogm_parse_chain):
	* ext/theora/theoradec.c: (gst_theora_dec_init), (_inc_granulepos),
	(theora_dec_src_convert), (theora_dec_sink_convert),
	(theora_dec_src_query), (theora_dec_sink_query),
	(theora_dec_src_event), (theora_dec_sink_event),
	(theora_handle_comment_packet), (theora_handle_type_packet),
	(theora_handle_header_packet), (theora_handle_data_packet),
	(theora_dec_chain):
	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(vorbis_dec_convert), (vorbis_dec_src_query),
	(vorbis_dec_sink_query), (vorbis_dec_src_event),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet), (vorbis_handle_header_packet),
	(copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
	(gst_vorbisenc_sink_query), (gst_vorbisenc_init),
	(gst_vorbisenc_sink_event), (gst_vorbisenc_chain):
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_query):
	* gst/playback/test3.c: (update_scale):
	* gst/sine/gstsinesrc.c: (gst_sinesrc_setcaps),
	(gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start):
	* gst/subparse/gstsubparse.c: (gst_subparse_init):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
	(gst_videotestsrc_src_query):
	* gst/videotestsrc/videotestsrc.c: (paint_hline_I420),
	(paint_hline_Y41B), (paint_hline_Y42B), (paint_hline_Y800),
	(paint_hline_YUV9):
	* sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
	Port to new query API.
	Updated seek.
	Cleanups in x[v]imagesink

2005-05-09  Andy Wingo  <wingo@pobox.com>

	* ext/alsa/gstalsasink.h:
	* ext/gnomevfs/gstgnomevfssrc.c:
	(gst_gnomevfssrc_get_icy_metadata):
	* ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek)
	(gst_ogg_demux_read_chain, gst_ogg_demux_read_end_chain)
	* ext/theora/theoradec.c (theora_dec_src_query)
	(theora_dec_src_event, theora_dec_sink_event)
	(theora_handle_comment_packet, theora_handle_data_packet):
	* ext/theora/theoraenc.c (theora_enc_chain):
	* ext/vorbis/vorbisdec.c (vorbis_dec_src_event)
	(vorbis_dec_sink_event, vorbis_handle_comment_packet):
	* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
	* gst/typefind/gsttypefindfunctions.c (mp3_type_find)
	(qt_type_find):
	* gst/videotestsrc/videotestsrc.c (paint_setup_I420)
	(paint_setup_YV12, paint_setup_YUY2, paint_setup_UYVY)
	(paint_setup_YVYU, paint_setup_IYU2, paint_setup_Y41B)
	(paint_setup_Y42B, paint_setup_Y800, paint_setup_IMC1)
	(paint_setup_IMC2, paint_setup_IMC3, paint_setup_IMC4)
	(paint_setup_YVU9, paint_setup_YUV9, paint_setup_xRGB8888)
	(paint_setup_xBGR8888, paint_setup_RGBx8888)
	(paint_setup_BGRx8888, paint_setup_RGB888, paint_setup_BGR888)
	(paint_setup_RGB565, paint_setup_xRGB1555):
	* gst/videotestsrc/videotestsrc.h:
	* sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc):
	* sys/xvimage/xvimagesink.c (gst_xvimagesink_get_xv_support)
	(gst_xvimagesink_setcaps, gst_xvimagesink_buffer_alloc):
	GCC4 fixes.
	
	* ext/ogg/gstoggdemux.c (gst_ogg_demux_find_chains): Use the new
	gst_pad_query_position. Fixes oggdemux.

2005-05-08  David Schleef  <ds@schleef.org>

	* configure.ac: Require liboil.
	* gst/videotestsrc/gstvideotestsrc.c: Fix up liboil calls, add
	a few more.
	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:

2005-05-06  Wim Taymans  <wim@fluendo.com>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
	Well, unreffing a buffer right before pushing it is asking
	for trouble..

2005-05-06  Christian Schaller  <uraeus@gnome.org>

	* pkgconfig/gstreamer-libs.pc.in: add missing library calls

2005-05-06  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
	(gst_audio_convert_fixate), (gst_audio_convert_channels):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
	(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
	* gst/sine/Makefile.am:
	* gst/sine/gstsinesrc.c: (gst_sinesrc_get_type),
	(gst_sinesrc_class_init), (gst_sinesrc_init),
	(gst_sinesrc_src_fixate), (gst_sinesrc_setcaps),
	(gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start),
	(gst_sinesrc_update_freq):
	* gst/sine/gstsinesrc.h:
	* gst/tcp/gstmultifdsink.c:
	* sys/xvimage/xvimagesink.c:
	Fixed negotiation wrt _peer_get_caps()
	Some cleanups.


2005-05-06  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
	(gst_audiosink_create_ringbuffer):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
	(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
	(gst_baseaudiosink_set_property), (build_linear_format),
	(debug_spec_caps), (debug_spec_buffer),
	(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
	(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
	(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
	(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
	(gst_ringbuffer_play), (gst_ringbuffer_pause),
	(gst_ringbuffer_stop), (gst_ringbuffer_delay),
	(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
	(wait_segment), (gst_ringbuffer_commit),
	(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
	(gst_ringbuffer_clear):
	* gst-libs/gst/audio/gstringbuffer.h:
	Make the base audiosink return an error when there is no
	audiobuffer negotiated.

2005-05-06  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* ext/Makefile.am:
	Disable cdparanoia until someone ports it!

2005-05-06  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
	(gst_ogg_demux_sink_activate):
	And revert after wingo's revert.. sigh..

2005-05-05  Andy Wingo  <wingo@pobox.com>

	* gst/audiorate/gstaudiorate.c (gst_audiorate_class_init): Pacify
	GObject.
	* configure.ac: Return audiorate and subparse from the ghetto.
	Re-enable -Wall -Werror.
	* gst/subparse/gstsubparse.c:
	* gst/subparse/gstsubparse.h: Port to 0.9. Can operate loop-based
	or chain-based. Cleaned up a bit. Not tested.
	
2005-05-05  Christian Schaller <christian@fluendo.com> 

	* Makefile.am: remove stuff that is not building
	* configure.ac: remove stuff that is not building
	* examples/Makefile.am: remove stuff that is not building
	* ext/alsa/gstalsasink.c: add alsa/ before the alsalib.h file
	* ext/alsa/gstalsasink.h: add alsa/ before the alsalib.h file
	* sys/Makefile.am: remove stuff that is not building
	* testsuite/Makefile.am: remove stuff that is not building

2005-05-05  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_from_vorbiscomment_buffer), (gst_vorbis_tag_chain):
	* gst/adder/gstadder.h:
	* gst/audioconvert/gstchannelmix.c:
	(gst_audio_convert_fill_one_other):
	* gst/audiorate/gstaudiorate.c: (gst_audiorate_setcaps),
	(gst_audiorate_init), (gst_audiorate_chain):
	* gst/playback/gstplaybasebin.c: (setup_source):
	* gst/playback/test3.c: (update_scale):
	Some GCC4 fixes
	
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po: Foo

2005-05-05  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
	(gst_audio_convert_change_state), (gst_audio_convert_channels):
	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
	(gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
	(gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
	(gst_videotestsrc_init), (gst_videotestsrc_loop):
	Don't ignore _push() return values.
	Make sure no processing is done when shutting down.
	Videotestsrc pad activation fix.

2005-05-05  Wim Taymans  <wim@fluendo.com>

	* gst/adder/Makefile.am:
	* gst/adder/gstadder.c: (gst_adder_setcaps),
	(gst_adder_class_init), (gst_adder_init),
	(gst_adder_request_new_pad), (gst_adder_collected),
	(gst_adder_change_state):
	* gst/adder/gstadder.h:
	Ported adder as an example of a mixer element using
	collect pads. Needs more negotiation work.

2005-05-05  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (_inc_granulepos),
	(theora_dec_src_event), (theora_dec_sink_event),
	(theora_handle_comment_packet), (theora_handle_type_packet),
	(theora_handle_header_packet), (theora_handle_data_packet),
	(theora_dec_chain):
	* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
	(gst_theora_enc_init), (theora_enc_sink_setcaps),
	(theora_push_buffer), (theora_push_packet),
	(theora_enc_sink_event), (theora_enc_chain),
	(theora_enc_change_state), (theora_enc_set_property),
	(theora_enc_get_property):
	Added stream lock to decoder so that we can serialize
	the discont event.
	More theoraenc porting, recover from errors, do clean
	shutdown.

2005-05-05  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/README:
	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_sink_activate),
	(gst_ogg_print):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
	(gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
	(gst_ogg_mux_push_page), (gst_ogg_mux_queue_pads),
	(gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
	(gst_ogg_mux_send_headers), (gst_ogg_mux_collected),
	(gst_ogg_mux_change_state):
	Ported ogg muxer.

2005-05-05  Wim Taymans  <wim@fluendo.com>

	* docs/design-audiosinks.txt:
	* gst-libs/gst/audio/TODO:
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
	(gst_audiosink_create_ringbuffer):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
	(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
	(gst_baseaudiosink_set_property), (build_linear_format),
	(debug_spec_caps), (debug_spec_buffer),
	(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
	(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
	(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
	(gst_ringbuffer_release), (gst_ringbuffer_play),
	(gst_ringbuffer_pause), (gst_ringbuffer_stop),
	(gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
	(gst_ringbuffer_set_sample), (wait_segment),
	(gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
	(gst_ringbuffer_advance), (gst_ringbuffer_clear):
	More work on the audiosink, mostly debugging and a race in
	shutdown.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_sink_activate):
	* ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
	(vorbis_dec_src_query), (vorbis_dec_src_event),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet), (vorbis_handle_header_packet),
	(copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain):
	Don't crap out when seeking back to position 0.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
	(make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
	(make_vorbis_pipeline), (make_vorbis_theora_pipeline),
	(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
	(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline):
	Make audio sink configurable, use alsasink as default.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet), (vorbis_handle_header_packet),
	(copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain),
	(vorbis_dec_change_state):
	* ext/vorbis/vorbisdec.h:
	Refactor, use STREAM_LOCK.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (_inc_granulepos),
	(theora_dec_sink_event), (theora_handle_comment_packet),
	(theora_handle_type_packet), (theora_handle_header_packet),
	(theora_handle_data_packet), (theora_dec_chain),
	(theora_dec_change_state):
	Refactor a bit, use STREAM_LOCK.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/Makefile.am:
	* ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_get_caps),
	(gst_alsa_fixate_to_mimetype), (gst_alsa_fixate_field_nearest_int),
	(gst_alsa_link), (gst_alsa_close_audio):
	* ext/alsa/gstalsaplugin.c: (plugin_init):
	* ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
	(gst_alsasink_dispose), (gst_alsasink_base_init),
	(gst_alsasink_class_init), (gst_alsasink_init),
	(gst_alsasink_getcaps), (set_hwparams), (set_swparams),
	(alsasink_parse_spec), (gst_alsasink_open), (gst_alsasink_close),
	(xrun_recovery), (gst_alsasink_write), (gst_alsasink_delay),
	(gst_alsasink_reset):
	* ext/alsa/gstalsasink.h:
	Implement alsasink with simple open/write/close API. 
	Make alsa dir build by disabling compilation of code.

2005-04-28  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/audioclock.c:
	* gst-libs/gst/audio/audioclock.h:
	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_get_type),
	(gst_audio_clock_class_init), (gst_audio_clock_init),
	(gst_audio_clock_new), (gst_audio_clock_get_internal_time):
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
	(gst_audiosink_create_ringbuffer):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
	(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
	(gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property),
	(build_linear_format), (debug_spec_caps), (debug_spec_buffer),
	(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
	(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
	(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
	(gst_ringbuffer_init), (gst_ringbuffer_finalize),
	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
	(gst_ringbuffer_release), (gst_ringbuffer_play),
	(gst_ringbuffer_pause), (gst_ringbuffer_stop),
	(gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
	(gst_ringbuffer_set_sample), (wait_segment),
	(gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
	(gst_ringbuffer_advance), (gst_ringbuffer_clear):
	* gst-libs/gst/audio/gstringbuffer.h:
	Make ringbuffer faster and more simple by removing the locks
	in the playback thread.
	Add sample accurate playback based on buffer sample offsets.
	Make the baseaudiosink provide a clock.
	Parse caps in the base class.
	Correctly handle seeking, flushing and state changes.

2005-04-25  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* gst/audioconvert/Makefile.am:
	* gst/audioscale/Makefile.am:
	  Fix part of the build.  Come on guys, autogen didn't even work :)

2005-04-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/media-info/.cvsignore:
	* gst-libs/gst/media-info/Makefile.am:
	* gst-libs/gst/media-info/README:
	* gst-libs/gst/media-info/media-info-priv.c:
	* gst-libs/gst/media-info/media-info-priv.h:
	* gst-libs/gst/media-info/media-info-test.c:
	* gst-libs/gst/media-info/media-info.c:
	* gst-libs/gst/media-info/media-info.h:
	* gst-libs/gst/media-info/media-info.vcproj:
	* pkgconfig/Makefile.am:
	* pkgconfig/gstreamer-media-info-uninstalled.pc.in:
	* pkgconfig/gstreamer-media-info.pc.in:
	  Remove media-info, which is also successed by playbin (see Totem
	  implementation).

2005-04-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* configure.ac:
	* examples/Makefile.am:
	* examples/gstplay/.cvsignore:
	* examples/gstplay/Makefile.am:
	* examples/gstplay/player.c:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/play/.cvsignore:
	* gst-libs/gst/play/Makefile.am:
	* gst-libs/gst/play/play.c:
	* gst-libs/gst/play/play.h:
	* gst-libs/gst/play/play.vcproj:
	* pkgconfig/Makefile.am:
	* pkgconfig/gstreamer-play-uninstalled.pc.in:
	* pkgconfig/gstreamer-play.pc.in:
	  Remove libgstplay, playbin is now the official successor.

2005-04-25  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/xwindowlistener/Makefile.am:
	* gst-libs/gst/xwindowlistener/xwindowlistener.c:
	* gst-libs/gst/xwindowlistener/xwindowlistener.h:
	  Remove deprecated xwindowlistener (I've moved xwindowlistening
	  in the v4l/v4l2 plugins over to serverside).

2005-04-25  David Schleef  <ds@schleef.org>

	* examples/dynparams/Makefile.am: Move demo-dparams from gst/sine
	to examples/dynparams.  Examples do not belong interspersed with
	source code.
	* examples/dynparams/demo-dparams.c:
	* gst/sine/Makefile.am:
	* gst/sine/demo-dparams.c:

2005-04-25  David Schleef  <ds@schleef.org>

	Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/xwindowlistener/Makefile.am:

	Convert to 0.9 API, seems to work:
	* sys/ximage/Makefile.am:
	* sys/ximage/ximagesink.c:

2005-04-24  David Schleef  <ds@schleef.org>

	Link plugins against libraries:
	* ext/alsa/Makefile.am:
	* gst/tcp/Makefile.am:

	Remove asm code that should be in liboil
	* gst/videoscale/Makefile.am:
	* gst/videoscale/videoscale_x86_asm.s:

	gettext wants these checked in:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:

2005-04-24  David Schleef  <ds@schleef.org>

	Convert gst_main() to g_main_loop_run()
	* gst/playback/decodetest.c: (main):
	* gst/playback/test2.c: (main):
	* gst/playback/test3.c: (main):
	* gst/playback/test4.c: (main):

	Link plugins against libraries:
	* ext/libvisual/Makefile.am:
	* sys/xvimage/Makefile.am:

2005-04-24  David Schleef  <ds@schleef.org>

	* configure.ac: Remove idct and resample libs
	* gst-libs/gst/Makefile.am: same

	Remove usage of gst_library_load():
	* ext/alsa/gstalsaplugin.c: (plugin_init):
	* ext/libvisual/visual.c: (plugin_init):
	* ext/ogg/gstogg.c: (plugin_init):
	* ext/theora/theora.c: (plugin_init):
	* ext/vorbis/vorbis.c: (plugin_init):
	* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
	* gst/audioscale/gstaudioscale.c:
	* gst/adder/gstadder.c: (plugin_init):
	* gst/audioconvert/plugin.c: (plugin_init):
	* sys/ximage/ximagesink.c: (plugin_init):
	* sys/xvimage/xvimagesink.c: (plugin_init):
	* gst/tcp/gsttcpplugin.c: (plugin_init):

	Link plugins against libraries:
	* ext/ogg/Makefile.am:
	* ext/theora/Makefile.am:
	* ext/vorbis/Makefile.am:
	* gst/audioconvert/Makefile.am:

	Create proper libraries:
	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/video/Makefile.am:

	Move resample library to audioscale plugin directory:
	* gst-libs/gst/resample/Makefile.am:
	* gst-libs/gst/resample/README:
	* gst-libs/gst/resample/dtof.c:
	* gst-libs/gst/resample/dtos.c:
	* gst-libs/gst/resample/functable.c:
	* gst-libs/gst/resample/private.h:
	* gst-libs/gst/resample/resample.c:
	* gst-libs/gst/resample/resample.h:
	* gst-libs/gst/resample/resample.vcproj:
	* gst-libs/gst/resample/test.c:
	* gst/audioscale/Makefile.am:
	* gst/audioscale/README:
	* gst/audioscale/dtof.c:
	* gst/audioscale/dtos.c:
	* gst/audioscale/functable.c:
	* gst/audioscale/private.h:
	* gst/audioscale/resample.c:
	* gst/audioscale/resample.h:
	* gst/audioscale/test.c:

	Move tagedit library to gst-libs:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/tag/gstid3tag.c:
	* gst-libs/gst/tag/gsttagediting.c:
	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/gstvorbistag.c:
	* gst/tags/Makefile.am:
	* gst/tags/gstid3tag.c:
	* gst/tags/gstvorbistag.c:

	Fix for core changes:
	* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
	(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
	(gst_sinesrc_getrange):

2005-04-23  David Schleef  <ds@schleef.org>

	* gst-libs/gst/Makefile.am: Remove idct.  It hasn't been used
	in gst-plugins in a long time, and properly belongs in liboil.
	* gst-libs/gst/idct/Makefile.am:
	* gst-libs/gst/idct/README:
	* gst-libs/gst/idct/dct.h:
	* gst-libs/gst/idct/doieee:
	* gst-libs/gst/idct/fastintidct.c:
	* gst-libs/gst/idct/floatidct.c:
	* gst-libs/gst/idct/idct.c:
	* gst-libs/gst/idct/idct.h:
	* gst-libs/gst/idct/idtc.vcproj:
	* gst-libs/gst/idct/ieeetest.c:
	* gst-libs/gst/idct/intidct.c:

2005-04-20  Wim Taymans  <wim@fluendo.com>

	* docs/design-audiosinks.txt:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/TODO:
	* gst-libs/gst/audio/gstaudiosink.c:
	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
	(gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize),
	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
	(gst_audioringbuffer_delay), (gst_audiosink_base_init),
	(gst_audiosink_class_init), (gst_audiosink_init),
	(gst_audiosink_create_ringbuffer):
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init),
	(gst_baseaudiosink_init), (gst_baseaudiosink_set_property),
	(gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps),
	(gst_baseaudiosink_get_times), (gst_baseaudiosink_event),
	(gst_baseaudiosink_preroll), (gst_baseaudiosink_render),
	(gst_baseaudiosink_create_ringbuffer),
	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
	(gst_ringbuffer_class_init), (gst_ringbuffer_init),
	(gst_ringbuffer_dispose), (gst_ringbuffer_finalize),
	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
	(gst_ringbuffer_release), (gst_ringbuffer_play_unlocked),
	(gst_ringbuffer_play), (gst_ringbuffer_pause),
	(gst_ringbuffer_resume), (gst_ringbuffer_stop),
	(gst_ringbuffer_callback), (gst_ringbuffer_delay),
	(gst_ringbuffer_played_samples), (gst_ringbuffer_commit),
	(gst_ringbuffer_prepare_read), (gst_ringbuffer_clear):
	* gst-libs/gst/audio/gstringbuffer.h:
	An attempt at a set of audio base classes together with some
	design docs.

2005-04-20  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
	(gst_audio_convert_channels):
	Link against audio libs.
	Fix audio convert plugin.

2005-04-20  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter),
	(gst_ogg_demux_sink_activate):
	* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
	(theora_set_header_on_caps), (theora_enc_sink_event),
	(theora_enc_chain):
	Fix theora encoder.

2005-04-12  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter):
	* gst/playback/gstdecodebin.c: (find_compatibles):
	  Work with staticpadtemplates in elementfactories.

2005-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/playback/README:
	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	(compare_ranks), (print_feature), (gst_decode_bin_init),
	(dynamic_create), (dynamic_free), (find_compatibles),
	(mimetype_is_raw), (close_pad_link), (got_redirect),
	(try_to_link_1), (get_our_ghost_pad), (remove_element_chain),
	(new_pad), (no_more_pads), (unlinked), (close_link), (type_found),
	(gst_decode_bin_change_state):
	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (group_destroy), (group_commit),
	(check_queue), (queue_overrun), (queue_threshold_reached),
	(queue_out_of_data), (gen_preroll_element), (unknown_type),
	(new_decoded_pad), (setup_subtitle), (gen_source_element),
	(got_redirect), (setup_source), (play_base_eos),
	(gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
	(gst_play_base_bin_remove_element):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_init), (gst_play_bin_dispose),
	(gst_play_bin_set_property), (gen_video_element),
	(gen_text_element), (gen_audio_element), (remove_sinks),
	(gst_play_bin_send_event):
	* gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
	(stream_info_change_state), (gst_stream_info_set_mute):
	* gst/playback/gststreamselector.c: (gst_stream_selector_init),
	(gst_stream_selector_get_caps), (gst_stream_selector_setcaps),
	(gst_stream_selector_request_new_pad), (gst_stream_selector_event),
	(gst_stream_selector_chain):
	* gst/playback/test.c: (gen_video_element), (gen_audio_element),
	(main):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps),
	(gst_xvimagesink_setcaps), (gst_xvimagesink_get_times),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_chain),
	(gst_xvimagesink_buffer_alloc), (gst_xvimagesink_class_init):
	Raw and crude port of decodebin. 
	Make playbin compile.

2005-04-06  Wim Taymans  <wim@fluendo.com>

	* ext/gnomevfs/Makefile.am:
	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
	(gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
	(gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
	(gst_gnomevfssrc_create), (gst_gnomevfssrc_is_seekable),
	(gst_gnomevfssrc_get_size), (gst_gnomevfssrc_start),
	(gst_gnomevfssrc_stop):
	* ext/ogg/Makefile.am:
	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_get_data),
	(gst_ogg_demux_find_chains), (gst_ogg_demux_sink_activate):
	* ext/theora/Makefile.am:
	* ext/theora/theoradec.c: (_inc_granulepos),
	(theora_dec_sink_event), (theora_dec_chain):
	* ext/vorbis/Makefile.am:
	* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	(vorbis_dec_sink_event), (vorbis_dec_chain):
	* gst-libs/gst/audio/Makefile.am:
	* sys/xvimage/Makefile.am:
	Make gnomevfssrc extend the source base class.
	Fix linking against libs in various plugins.

2005-04-06  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/video/Makefile.am (libgstvideo_la_LDFLAGS): Use
	GST_BASE_LIBS.

	* configure.ac: Add check and AC_SUBST for libgstbase.

2005-03-31  Wim Taymans  <wim@fluendo.com>

	* examples/seeking/Makefile.am:
	* examples/seeking/cdparanoia.c: (main):
	* examples/seeking/cdplayer.c: (update_scale), (stop_seek),
	(play_cb), (pause_cb), (stop_cb), (main):
	* examples/seeking/playbin.c:
	* examples/seeking/seek.c: (dynamic_link), (make_mod_pipeline),
	(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
	(make_sid_pipeline), (make_vorbis_pipeline),
	(make_theora_pipeline), (make_vorbis_theora_pipeline),
	(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
	(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
	(make_playerbin_pipeline), (update_scale), (end_scrub), (do_seek),
	(seek_cb), (start_seek), (stop_seek), (play_cb), (pause_cb),
	(stop_cb), (main):
	* examples/seeking/spider_seek.c:
	* examples/seeking/vorbisfile.c:
	* ext/gnomevfs/Makefile.am:
	* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_base_init),
	(gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
	(gst_gnomevfssrc_get_property), (gst_gnomevfssrc_get),
	(gst_gnomevfssrc_open_file), (gst_gnomevfssrc_close_file),
	(gst_gnomevfssrc_getrange), (gst_gnomevfssrc_loop),
	(gst_gnomevfssrc_activate), (gst_gnomevfssrc_change_state),
	(gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
	* ext/ogg/README:
	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_get_type),
	(gst_ogg_pad_class_init), (gst_ogg_pad_init),
	(gst_ogg_pad_dispose), (gst_ogg_pad_finalize),
	(gst_ogg_pad_formats), (gst_ogg_pad_event_masks),
	(gst_ogg_pad_query_types), (gst_ogg_pad_getcaps),
	(gst_ogg_pad_src_convert), (gst_ogg_pad_src_query),
	(gst_ogg_pad_event), (gst_ogg_pad_reset),
	(gst_ogg_demux_factory_filter), (compare_ranks),
	(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
	(gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
	(gst_ogg_chain_new), (gst_ogg_chain_free),
	(gst_ogg_chain_new_stream), (gst_ogg_chain_get_stream),
	(gst_ogg_chain_has_stream), (gst_ogg_demux_base_init),
	(gst_ogg_demux_class_init), (gst_ogg_demux_init),
	(gst_ogg_demux_finalize), (gst_ogg_demux_handle_event),
	(gst_ogg_demux_submit_buffer), (gst_ogg_demux_seek),
	(gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
	(gst_ogg_demux_get_prev_page),
	(gst_ogg_demux_deactivate_current_chain),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_bisect_forward_serialno),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
	(gst_ogg_demux_find_pad), (gst_ogg_demux_find_chain),
	(gst_ogg_demux_find_chains), (gst_ogg_demux_chain_unlocked),
	(gst_ogg_demux_chain), (gst_ogg_demux_send_eos),
	(gst_ogg_demux_loop), (gst_ogg_demux_sink_activate),
	(gst_ogg_demux_change_state), (gst_ogg_print):
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
	(gst_ogg_mux_init), (gst_ogg_mux_sinkconnect),
	(gst_ogg_mux_next_buffer), (gst_ogg_mux_buffer_from_page),
	(gst_ogg_mux_push_page), (gst_ogg_mux_send_headers),
	(gst_ogg_mux_loop):
	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	* ext/theora/theoradec.c: (gst_theora_dec_init), (_inc_granulepos),
	(theora_dec_src_convert), (theora_dec_sink_convert),
	(theora_dec_src_query), (theora_dec_src_event),
	(theora_dec_sink_event), (theora_dec_chain),
	(theora_dec_change_state):
	* ext/theora/theoraenc.c: (gst_theora_enc_init),
	(theora_enc_sink_setcaps), (theora_buffer_from_packet),
	(theora_push_buffer), (theora_enc_sink_event), (theora_enc_chain),
	(theora_enc_change_state):
	* ext/vorbis/Makefile.am:
	* ext/vorbis/oggvorbisenc.c:
	* ext/vorbis/oggvorbisenc.h:
	* ext/vorbis/vorbis.c: (plugin_init):
	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(vorbis_dec_src_query), (vorbis_dec_src_event),
	(vorbis_dec_sink_event), (vorbis_dec_chain),
	(vorbis_dec_change_state):
	* ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init),
	(gst_vorbisenc_sink_setcaps), (gst_vorbisenc_init),
	(gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
	(gst_vorbisenc_sink_event), (gst_vorbisenc_chain),
	(gst_vorbisenc_change_state):
	* ext/vorbis/vorbisenc.h:
	* ext/vorbis/vorbisparse.c: (vorbis_parse_chain):
	* gst-libs/gst/audio/audioclock.c:
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
	(gst_audiofilter_init), (gst_audiofilter_chain):
	* gst-libs/gst/audio/testchannels.c: (main):
	* gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
	* gst-libs/gst/media-info/media-info-priv.c: (gmip_reset),
	(gmip_find_type), (gmip_find_stream), (gmip_find_track_metadata),
	(gmip_find_track_streaminfo), (gmip_find_track_format):
	* gst-libs/gst/media-info/media-info.c:
	(gst_media_info_read_idler):
	* gst-libs/gst/play/play.c: (gst_play_get_sink_element),
	(gst_play_get_all_by_interface):
	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
	(gst_riff_parse_chunk), (gst_riff_parse_file_header),
	(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
	(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
	(gst_riff_parse_info):
	* gst-libs/gst/riff/riff-read.h:
	* gst-libs/gst/riff/riff.c: (plugin_init):
	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/gstvideosink.c: (gst_videosink_init),
	(gst_videosink_class_init), (gst_videosink_get_type):
	* gst-libs/gst/video/videosink.h:
	* gst/audioconvert/bufferframesconvert.c:
	(buffer_frames_convert_init), (buffer_frames_convert_fixate),
	(buffer_frames_convert_setcaps), (buffer_frames_convert_chain):
	* gst/audioconvert/channelmixtest.c: (main):
	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
	(gst_audio_convert_chain),
	(gst_audio_convert_caps_remove_format_info),
	(gst_audio_convert_getcaps), (gst_audio_convert_parse_caps),
	(gst_audio_convert_setcaps), (_fixate_caps_to_int),
	(gst_audio_convert_fixate), (gst_audio_convert_get_buffer),
	(gst_audio_convert_buffer_to_default_format),
	(gst_audio_convert_buffer_from_default_format),
	(gst_audio_convert_channels):
	* gst/audioconvert/gstchannelmix.h:
	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_caps_remove_format_info), (gst_ffmpegcsp_getcaps),
	(gst_ffmpegcsp_configure_context), (gst_ffmpegcsp_setcaps),
	(gst_ffmpegcsp_init), (gst_ffmpegcsp_chain):
	* gst/tags/gstid3tag.c: (gst_tag_extract_id3v1_string):
	* gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
	* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
	(mp3_type_find), (mpeg2_sys_type_find), (mpeg1_sys_type_find),
	(mpeg_video_type_find), (mpeg_video_stream_type_find),
	(dv_type_find):
	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_videotestsrc_class_init), (gst_videotestsrc_src_negotiate),
	(gst_videotestsrc_src_link), (gst_videotestsrc_parse_caps),
	(gst_videotestsrc_src_accept_caps), (gst_videotestsrc_setcaps),
	(gst_videotestsrc_src_unlink), (gst_videotestsrc_activate),
	(gst_videotestsrc_change_state), (gst_videotestsrc_getcaps),
	(gst_videotestsrc_init), (gst_videotestsrc_src_query),
	(gst_videotestsrc_handle_src_event), (gst_videotestsrc_loop):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_fixate),
	(gst_xvimagesink_getcaps), (gst_xvimagesink_setcaps),
	(gst_xvimagesink_change_state), (gst_xvimagesink_get_times),
	(gst_xvimagesink_show_frame), (gst_xvimagesink_chain),
	(gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_navigation_send_event),
	(gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_expose),
	(gst_xvimagesink_set_property), (gst_xvimagesink_finalize),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init):
	* sys/xvimage/xvimagesink.h:
	Plugin port to 0.9, ogg/theora playback should work in the seek
	example now.
	Removed old examples.
	Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
	explained in 0.9 TODO doc.


2005-02-23  Thomas Vander Stichele  <thomas at apestaart dot org>

	* autogen.sh:
	* configure.ac:
	* ext/Makefile.am:
	* gst/Makefile.am:
	* po/POTFILES.in:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* sys/Makefile.am:
	* testsuite/Makefile.am:
	  remove a whole bunch of plugins.  This module now contains a set
	  of free reference plugins/elements as agreed.

2005-02-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  hunting season on 0.9 is now OPEN
