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2008-10-08  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Daniel Drake <dsd at laptop dot org>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
	Unref all buffers when clearing collectpads. Fixes bug #546955.

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2008-10-08  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Based on a patch by: Klaas <klaas at rivercrew dot net>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	* ext/vorbis/vorbisenc.h:
	Keep track of the upstream segments and use the running time on that
	segment instead of the buffer timestamp everywhere. Fixes bug #525807.

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2008-10-08  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/audioconvert/audioconvert.c: (audio_convert_convert):
	Prevent overflows with big buffer when calculating the size of
	the intermediate buffer by using gst_util_uint64_scale() instead of
	plain arithmetics. Fixes bug #552801.

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2008-10-08  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Pavel Zeldin <pzeldin at gmail dot com>

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
	(gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
	(gst_clock_overlay_init), (gst_clock_overlay_set_property),
	(gst_clock_overlay_get_property):
	* ext/pango/gstclockoverlay.h:
	API: Add ability to specify format for date/time display by
	adding a "time-format" property.
	Fixes bug #554879.

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2008-10-08  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Jan Gerber <j at oil21 dot org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	(gst_riff_create_video_template_caps):
	Add FFV1 fourcc to support playback of FFMPEG lossless video
	in AVI. Fixes bug #555319.

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2008-10-08  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Håvard Graff <havard dot graff at tandberg dot com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	Implement skew clock slaving. Fixes #552559.

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2008-10-08  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/multichannel.c:
	* gst-libs/gst/audio/testchannels.c:
	Fix include of config.h

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2008-10-06  Wim Taymans  <wim.taymans@collabora.co.uk>

	Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>

	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
	(print_media), (gst_sdp_message_dump):
	Fix parsing of the c= field containing multicast addresses.
	Fixes #552199.
	Add the connection info to the session or streams.
	Fix parsing of the bandwidth.
	Add debugging for the connections and bandwidths for a media.
	Add debugging for the bandwidth of the session. 

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2008-10-06  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_change_state):
	Configure the next seqnum and timestamp in the state change so that they
	can be queried soon after.

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2008-10-06  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain):
	Improve debugging of the rtptime.

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2008-10-05  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	Back to development -> 0.10.21.1

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2008-10-05  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c: (flac_type_find),
	(plugin_init):
	Add typefinding for FLAC without headers in the beginning.
	This is already handled by flacdec and friends.

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2008-10-05  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
	(plugin_init):
	Add typefinder for MXF.

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2008-10-03  Jan Schmidt  <jan.schmidt@sun.com>

	* tests/icles/Makefile.am:
	Only build test-colorkey if GTK+ is available.

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=== release 0.10.21 ===

2008-10-03  Jan Schmidt <jan.schmidt@sun.com>

	* configure.ac:
	  releasing 0.10.21, "We Could Be Wrong"

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2008-09-28  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	0.10.20.4 pre-release

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2008-09-25  Wim Taymans  <wim.taymans@collabora.co.uk>

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	Patch by: Vincent Penquerc'h <ogg dot k dot ogg dot k at googlemail dot com>
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	* ext/theora/theoraparse.c: (theora_parse_set_streamheader):
	Set the BOS flag on the BOS packet. Fixes #553244.

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2008-09-23  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtsp/gstrtspmessage.c:
	(gst_rtsp_message_parse_request),
	(gst_rtsp_message_parse_response):
	Fix the g_return_val_if_fail() statements.

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2008-09-22  Michael Smith <msmith@songbirdnest.com>

	* gst-libs/gst/tag/gsttagdemux.c:
	  Fail to activate if there's insufficient data in the file to be usable,
	  preventing an assertion fail later. Fixes #552960

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2008-09-15  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/pbutils/descriptions.c:
	* gst/typefind/gsttypefindfunctions.c:
	  Recognise Kate subtitle streams (#550582).

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2008-09-13  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
	  Remove trailing comma from enum list, which causes problems
	  with -pendantic (#550729).

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2008-09-10  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	0.10.20.2 pre-release

	* po/LINGUAS:
	* po/id.po:
	* po/pt_BR.po:

	New translations.

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2008-09-05  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/interfaces/propertyprobe.c:
	  (gst_property_probe_get_properties),
	  (gst_property_probe_get_property),
	  (gst_property_probe_probe_property),
	  (gst_property_probe_probe_property_name),
	  (gst_property_probe_needs_probe),
	  (gst_property_probe_needs_probe_name),
	  (gst_property_probe_get_values),
	  (gst_property_probe_get_values_name),
	  (gst_property_probe_probe_and_get_values),
	  (gst_property_probe_probe_and_get_values_name):
	  More sanity checks for our second-favourite interface.

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2008-09-05  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/propertyprobe.c:
	  Check for NULL pointer, in the hope that this fixes #532864.

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2008-09-05  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
	  No really, the next release is 0.10.21 (fix Since: tags in docs).

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2008-09-04  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
	Disable a code path that is now called but causes a deadlock for some
	reason and is unneeded.

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2008-09-04  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c:
	* sys/xvimage/xvimagesink.h:
	  Add a "draw-border" property that can be set to false to disable
	  drawing borders.
	  
	* tests/icles/test-colorkey.c:
	* tests/icles/Makefile.am:
	  Add new test application for the colorkey handling.

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2008-09-03  Edward Hervey  <edward.hervey@collabora.co.uk>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
	This will also be fixed for upcoming gst-ffmpeg release so that once
	this release of -base is out, it will work with the latest gst-ffmpeg
	release.

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2008-09-03  Edward Hervey  <edward.hervey@collabora.co.uk>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Add Truespeech mapping for RIFF formats (AVI/WAV).
	Fixes #550656

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2008-09-03  Mark Nauwelaerts  <mark.nauwelaerts@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
	Fixes #550638.

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2008-09-03  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/subparse/Makefile.am:
	* gst/subparse/gstsubparse.c:
	* gst/subparse/samiparse.c:
	* tests/check/elements/subparse.c:
	  Rework last change, so that we build subparse, but just disable the
	  sami parse functionality, if we're configured to not use xml. In the
	  tests only the sami test is disabled now.

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2008-09-02  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Disable subparse when xml is disabled. It woundn't work anyway. Fixes
	  test runs.

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2008-09-02  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* po/POTFILES.in:
	  Add some more files with strings for translation.

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2008-09-02  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/tag/gstvorbistag.c:
	* tests/check/libs/tag.c:
	  Use new geo location tags from core. Fixes #481169

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2008-09-01  Edward Hervey  <edward.hervey@collabora.co.uk>

	* tests/check/elements/audioresample.c: (setup_audioresample),
	(fail_unless_perfect_stream), (test_perfect_stream_instance),
	(test_discont_stream_instance):
	Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
	Add debugging for coherence.

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2008-08-30  Tim-Philipp Müller  <tim.muller at collabora co uk>

	Patch by: Jonathan Matthew  <notverysmart gmail com>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Add typefinder for PDF documents (which is nice to have, since it's a
	  common format, but also helps prevent false positives). Fixes #549814.

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2008-08-27  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
	(no_more_pads_cb):
	Fix nasty race where multiple decodebins could start pushing data before
	we manage to configure the sinks, resulting in not-linked errors in
	typical RTSP streaming cases.

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2008-08-26  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
	Since we now call stop, we trigger this code path that causes a deadlock
	is apparently not needed.

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2008-08-26  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
	(gst_ring_buffer_stop):
	Also allow the case where the ringbuffer was paused when we try to stop
	it so that the basesrc stop function is still called.

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2008-08-23  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Mike Ruprecht <cmaiku at gmail dot com>

	* sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
	Reprobe devices again instead of taking a cached list as new
	devices could've been plugged in. Fixes bug #549062.

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2008-08-23  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Alessandro Dessina <alessandro nnva org>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
	(gst_ogg_demux_activate_chain):
	Don't add pads and activate them for skeleton streams. These are already
	handled inside oggdemux. Fixes bug #537599.

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2008-08-22  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
	Reset variable so that query and convert fail after going back to
	READY. Fixes #548898.

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2008-08-22  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	If a buffer arrives with a timestamp before the timestamp+duration
	of the previous buffer clip it instead of dropping it completely.
	Slight improvement for the unfixable bug #548913.

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2008-08-21  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	Take the current timestamp instead of timestamp+duration for the offset.
	This offset will later be used for calculating the timestamp and
	otherwise vorbisdec will interpolate timestamps wrong if upstream
	only sends timestamps and no granulepos.

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2008-08-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c:
	  Don't crash when having no visualisations.

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2008-08-16  David Schleef  <ds@schleef.org>

	* gst/typefind/gsttypefindfunctions.c: DV typefinding.  Remove
	  check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
	  Fixes #548065.

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2008-08-15  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
	When cleaning up the caps fields also remove "depth" for the same
	reason we remove "width".

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2008-08-14  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
	  Add Lead H.264 here as well.

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2008-08-14  Julien Moutte  <julien@fluendo.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	(gst_riff_create_video_template_caps): Add Lead H.264 variant.

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2008-08-13  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	When not slaved to another clock also subtract the base_time from our
	internal clock time to get the running time.

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2008-08-12  David Schleef  <ds@schleef.org>

	* ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
	  since it has no basis in libtheora.

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2008-08-12  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/propertyprobe.h:
	  Remove double "interface" from doc-string.

	* gst-libs/gst/interfaces/xoverlay.h:
	  Document interface.

	* gst-libs/gst/riff/riff.c:
	  Add basic doc blobs.

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2008-08-11  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/audio/Makefile.am:
	  Don't try to build that example anymore.

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2008-08-11  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/audio/.cvsignore:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	* gst-libs/gst/audio/make_filter:
	  Move audiofiltertemplate to gst-template.

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2008-08-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiosrc.h:
	  More docs and shuffling. What can we do with the hundreds of #defines.

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	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/interfaces/propertyprobe.h:
	* gst-libs/gst/tag/gsttagdemux.h:
	  Reducing number of dundocumented symbols.

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2008-08-11  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio.c:
	  Fix doc comment syntax.

	* gst-libs/gst/interfaces/propertyprobe.c:
	  Add more doc-comments and a FIXME: for the signal.	

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2008-08-07  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
	(gst_ogg_mux_request_new_pad):
	* ext/ogg/gstoggmux.h:
	Don't pretend to support NEWSEGMENT events, instead override the
	GstCollectPads event function to return FALSE on NEWSEGMENT events
	and do the normal work for other events.

	This prevents elements like flacenc to seek to the start and rewrite
	some data which then results in a broken Ogg packet.

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2008-08-07  Tim-Philipp Müller  <tim.muller at collabora co uk>

	Patch by: Frederic Crozat <fcrozat@mandriva.org>

	* ext/alsa/gstalsaplugin.c: (plugin_init):
	* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
	* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
	* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
	* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
	* gst/playback/gstdecodebin.c: (plugin_init):
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
	* gst/playback/gstplayback.c: (plugin_init):
	* gst/playback/gstqueue2.c: (plugin_init):
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
	* sys/v4l/gstv4l.c: (plugin_init):
	Make sure gettext returns translations in UTF-8 encoding rather
	than in the current locale encoding (#546822).

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2008-08-06  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/pbutils/descriptions.c:
	  Add audio/x-qdm for qtdemux.

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2008-08-05  Stefan Kost  <ensonic@users.sf.net>

	* ext/vorbis/vorbisdec.c:
	  Do not leak old taglist.

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2008-08-04  Stefan Kost  <ensonic@users.sf.net>

	* tests/icles/test-scale.c:
	  Include <stdlib.h> for atoi().

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2008-08-04  Andy Wingo  <wingo@pobox.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
	documentation fix.

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2008-08-01  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c:
	  Cleanup lots of empty lines that came from gst-indent going havoc
	  before I added the INDENT_ON/OFF marker some time agao.

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2008-08-01  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst-libs/gst/riff/riff-read.c:
	  Bump requirement to latest core and use new tag for riff formats.
	  Needed for #520694.

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2008-08-01  Wim Taymans  <wim.taymans@collabora.co.uk>

	* tests/examples/dynamic/Makefile.am:
	* tests/examples/dynamic/codec-select.c: (make_encoder),
	(make_pipeline), (do_switch), (my_bus_callback), (main):
	Add example app that dynamically switches between 3 'encoders'.

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2008-07-31  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
	Add some more comments.

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2008-07-31  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
	(gst_video_test_src_create):
	Discard buffers of the wrong size after renegotiation, this is perfectly
	possible with things like capsfilter that could suggest caps changes
	upstream without knowing the size of the buffer.

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	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-scale.c: (make_pipeline), (main):
	Add dynamic rescaling tests for the new basetransform.

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	* gst/audioconvert/Makefile.am:
	  Dist recently-added gstfastrandom.h.

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2008-07-30  Edward Hervey  <edward.hervey@collabora.co.uk>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
	Fix a "may be used uninitialized in this function" which weirdly only
	appears on macosx (?).

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2008-07-30  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	  Adding acid chunk for tempo and loop information.

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	* sys/xvimage/Makefile.am:
	  floor() needs linking to $(LIBM).

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2008-07-29  Stefan Kost  <ensonic@users.sf.net>

	* ext/gnomevfs/gstgnomevfssrc.c:
	  Aggregate short reads and add some comments and debug logging.
	  Fixes #537380

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2008-07-29  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gstplaybasebin.c:
	  Fix property doc markup (its not a signal).

	* sys/xvimage/xvimagesink.c:
	  Add since tag for new proeprties (also add sice tags fro the last two
	  other additions).

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2008-07-29  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c:
	* sys/xvimage/xvimagesink.h:
	  Add autofill/colorkey properties. Fixes #538656.

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	* sys/xvimage/xvimagesink.c:
	  Fix rounding errors when converting colorbalance values
	  between hardware and object property ranges.  Partial
	  fix for #537889, however, there still seems to be a small
	  drift problem that could be totem's fault.

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2008-07-28  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
	Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
	This fixes a critical warning.

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2008-07-28  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* ext/ogg/gstoggmux.c:
	Allow muxing of CELT into Ogg streams.

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2008-07-28  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
	(plugin_init):
	Add simple typefinder for the CELT codec (www.celt-codec.org).

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2008-07-27  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Jan Gerber <j at oil21 dot org>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
	Fix calculation of the start time from skeleton streams.
	Fixes bug #530068.

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2008-07-24  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c:
	  Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.

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2008-07-23  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	(gst_audio_quantize_setup_dither),
	(gst_audio_quantize_free_dither):
	* gst/audioconvert/gstfastrandom.h:
	Implement a linear congruential generator as pseudo random number
	generator for the dither noise. This is about 2 times faster than
	using GLib's mersenne twister. Also this uses only integer math for
	generating integers while GLib internally uses floating point math.

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2008-07-23  Michael Smith <msmith@songbirdnest.com>

	* configure.ac:
	  Remove AC_ISC_POSIX; it breaks on some systems and is not needed.

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2008-07-23  Tim-Philipp Müller  <tim.muller at collabora co uk>

	Patch by: Damien Lespiau  <damien.lespiau gmail com>

	* gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
	  Use GST_STR_NULL to avoid crashes with libcs that don't
	  like NULL strings in printf args (such as the win32 one).
	  Fixes #544306.

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2008-07-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
	Oops - set the size of the image used for probing back to 1x1, for
	consistency with ximagesink

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2008-07-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
	(gst_ximagesink_ximage_new):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
	(gst_xvimagesink_xvimage_new):

	Apparently on Solaris and OS/X (at least), it's not legal to ask the
	X server to attach to a shared memory segment after we've deleted it,
	with the result that MIT-SHM is disabled. Instead, remove it only after 
	X succeeds in attaching too.

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2008-07-16  David Schleef  <ds@schleef.org>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/audiotestsrc/gstaudiotestsrc.h:
	  Add 'ticks', a 1/30 second sine wave pulse every second.

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2008-07-15  David Schleef  <ds@schleef.org>

	* gst-libs/gst/video/video.c: Revert ABI change.

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2008-07-15  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	Make it impossible to have NULL caps at the point where we set
	framerate and other things. Also don't return immediately for "3ivd"
	video and let framerate, etc be set. Might fix bug #542508.

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2008-07-14  Mark Nauwelaerts  <mark.nauwelaerts@collabora.co.uk>

	* gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
	Video format can also be conveniently determined from (many)
	non-fixed caps.

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2008-07-14  Jan Schmidt  <thaytan@noraisin.net>

	* gst/playback/gstplaybasebin.c:
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c:
	* gst/playback/gststreamselector.c:
	First stab at integrating DVD subpicture overlay into
	playbin. Successfully plugs and plays, but the queues need
	shrinking - 3 seconds of video is too much buffering.

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2008-07-11  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	  Remove now obsolete note in the docs.

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2008-07-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-overrides.txt:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* docs/plugins/gst-plugins-base-plugins.interfaces:
	* docs/plugins/gst-plugins-base-plugins.prerequisites:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playback.xml:
	* docs/plugins/inspect/plugin-queue2.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-uridecodebin.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	* ext/gio/gstgiosink.c:
	* ext/gio/gstgiosrc.c:
	* ext/gio/gstgiostreamsink.c:
	* ext/gio/gstgiostreamsrc.c:
	* ext/gnomevfs/gstgnomevfssink.c:
	* ext/gnomevfs/gstgnomevfssrc.c:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/pango/gstclockoverlay.c:
	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextrender.c:
	* ext/pango/gsttimeoverlay.c:
	* ext/theora/theoradec.c:
	* ext/theora/theoraenc.c:
	* ext/theora/theoraparse.c:
	* ext/vorbis/vorbisdec.c:
	* ext/vorbis/vorbisenc.c:
	* ext/vorbis/vorbisparse.c:
	* ext/vorbis/vorbistag.c:
	* gst/adder/gstadder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	* gst/gdp/gstgdpdepay.c:
	* gst/gdp/gstgdppay.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstqueue2.c:
	* gst/playback/gsturidecodebin.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gsttcpserversink.c:
	* gst/videorate/gstvideorate.c:
	* gst/videoscale/gstvideoscale.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/volume/gstvolume.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  Cleanup Plugin docs. Link to signals and properties. Fix sub-section
	  titles. Drop mentining that all our example pipelines are "simple"
	  pipelines.

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2008-07-07  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* tests/examples/seek/Makefile.am:
	Fix out of tree build by adding all required CFLAGS.

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2008-07-07  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin.c: (add_raw_queue):
	And ref the pad before returning it again when linking to the queue
	failed. Otherwise we will unref the pad twice later and things break.

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2008-07-07  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/playback/gstdecodebin.c: (add_raw_queue):
	If linking the raw pad with a queue fails, try it without a queue
	instead of failing completely. This should never happen.

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2008-07-07  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>

	* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
	Add a queue after a demuxer if the demuxer outputs raw data. This was
	done before only for non-raw data but is required in this case too.
	Fixes bug #540215.

	decodebin2 doesn't have this issue because all streams of a group
	go through multiqueue.

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2008-07-03  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	Makes libgstsdp compile with mingw32 by defining the right WINVER so
	that getaddrinfo() can be used. Fixes #541358.

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2008-07-01  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_class_init), (gst_video_test_src_init),
	(gst_video_test_src_set_property),
	(gst_video_test_src_get_property), (gst_video_test_src_create):
	* gst/videotestsrc/gstvideotestsrc.h:
	Cleanups, use default property values as defines.
	Add property to enable/disable peer buffer allocation.

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2008-06-30  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
	* tests/check/pipelines/streamheader.c: (streamheader_suite):
	Enable unit tests on PPC again as the bugs are now fixed.

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2008-06-30  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
	Fixes bug #540351.

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2008-06-30  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
	it on other formats. Also adjust the unit size only for that format
	to not include the palette. Fixes bug #540497.

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	* gst/adder/gstadder.c:
	  Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.

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	* ChangeLog:
	  ChangeLog surgery.
  
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	* tests/examples/seek/seek.c:
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	  Move variable into ifdef too.
	

2008-06-27  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c:
	  Include config.h and check if we have X. Remove XInitThread(), don't
	  think its needed. Fixes: #540334.
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2008-06-26  Sebastian Dröge  <sebastian.droege@collabora.co.uk>

	Patch by: Sam Morris <sam at robots dot org to uk>

	* gst-libs/gst/interfaces/mixertrack.c:
	(gst_mixer_track_class_init), (gst_mixer_track_get_property),
	(gst_mixer_track_set_property):
	API: Add "index" property to GstMixerTrack to differantiate between
	multiple mixer tracks with the same label.

	* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	Set the "index" property of GstMixerTrack to the index given by ALSA.
	Fixes bug #528299.

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2008-06-25  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/Makefile.am:
	* tests/examples/seek/seek.c:
	  Remove libgstvideo usage. Use gtk_get_option_group instead of
	  gtk_init().

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	* tests/check/Makefile.am:
	  Name the test registry format neutral.

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	* gst/playback/gstqueue2.c:
	  Do not double notify. Remove the unsued return value.

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2008-06-24  Stefan Kost  <ensonic@users.sf.net>

	* ext/alsa/gstalsamixer.c:
	  Also consider "speaker" as a name for master volume. If that doesn't
	  help look for the first non-mono volume control that also has a
	  playback switch.

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2008-06-24  Stefan Kost  <ensonic@users.sf.net>

	* ChangeLog:
	  Forgot to save the ChangeLog :/

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	* tests/examples/seek/Makefile.am:
	* tests/examples/seek/seek.c:
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	  Embedd the xwindow. Allow to play multiple files.
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2008-06-24  Jan Schmidt  <jan.schmidt@sun.com>

	* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
	(gst_ximagesink_setcaps):
	* sys/ximage/ximagesink.h:
	When the caps change, make sure to re-draw borders in
	force-aspect-ratio=true mode.
	* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
	Don't clear the border_draw flag until we actually draw the border.
	* tests/check/Makefile.am:
	Ignore alsasink/src during the states test too, so it doesn't fail
	when running without access to the sound device.

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2008-06-22  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c:
	  Fix crasher when playing a parse-launch line the 2nd time.

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2008-06-21  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/pipelines/oggmux.c:
	  Properly ifdef tests to fix compilation.

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	* gst/playback/gstplay-marshal.list:
	* gst/playback/gstplaybin2.c:
	  Add get-video-pad, get-audio-pad, get-text-pad action signals to
	  playbin2. This allows the user to get to the selector's sinkpads, and
	  thus inspect a range of things - caps, tags, etc. 

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	* gst/playback/gstplaybin2.c:
	  Use a different constant for the convert-frame signal id.
	  Fixes #537009.

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	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaysink.c:
	  Fix a whole bunch of typos in comments and log statements. 

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2008-06-20  Michael Smith <msmith@songbirdnest.com>

	* sys/xvimage/xvimagesink.c:
	  Don't set colour balance values on the Xv port if the user hasn't
	  changed them (via properties or the interface). Avoids accumulating
	  rounding errors for the common case.
	  Partial fix for bug #537889.

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2008-06-20  Michael Smith <msmith@songbirdnest.com>

	* gst/playback/gstdecodebin2.c:
	  Ensure decodebin2 emits 'drained' signal once, and only once, when all
	  pads are drained.

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2008-06-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
	(gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
	Report the encoder latency. Fixes #538232.

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2008-06-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
	(notify_source), (activate_group):
	Implement the source property, emit notify when it changes in the
	underlying uridecodebin.

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2008-06-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* tests/examples/seek/seek.c: (stop_cb):
	Free and clear the seek element list so that we don't use invalid
	references when seeking after recreating a gst-launch line.

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2008-06-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_render):
	Report latency even if we are not live instead of hiding it.
	Take ts-offset and render-delay of the basesink into account when
	scheduling samples.
	Rework the clipping code so that we can take the various offsets into
	account and still do correct clipping.

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2008-06-20  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	Bump verion back to devel -> 0.10.20.1

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2008-06-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
	Don't increase the size of non-string image buffers by one as this
	might in theory confuse decoders. Still increase it by one for string
	image buffers to append '\0'.

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2008-06-20  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Antoine Tremblay <hexa00 at gmail dot com>
	
	* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
	Fix a buffer memleak and remove a confusing and wrong debug output.
	Fixes bug #538663.

Jan Schmidt's avatar
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=== release 0.10.20 ===

2008-06-18  Jan Schmidt <jan.schmidt@sun.com>

	* configure.ac:
	  releasing 0.10.20, "Here I Go Again"

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2008-06-11  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	0.10.19.3 pre-release

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2008-06-11  Jan Schmidt  <jan.schmidt@sun.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	Fix build on win32.
	Patch By: David Schleef <ds@schleef.org>
	Fixes: #536874

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2008-06-11  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
	(gst_gio_base_src_create):
	* ext/gio/gstgiobasesrc.h:
	Try to read the requested number of bytes, even if the first
	read returns less than requested, until nothing is read anymore
	or we have the requested amount of bytes. This fixes playback of
	files via Samba as Samba only allows to read 64k at once.

	Implement a caching algorithm that makes sure that we read at
	least 4k of data every time. Some elements will try to read a few
	bytes, then seek, read again a few bytes and so on and this is
	painfully slow as every operation has to go over DBus if GVfs is
	used as backend.

	Fixes bug #536849 and #536848.

	* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
	(gst_gio_src_check_get_range):
	Override check_get_range() to blacklist http/https URIs
	and whitelist file URIs. More to be added on demand.

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2008-06-05  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	0.10.19.2 pre-release

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2008-06-04  Jan Schmidt  <jan.schmidt@sun.com>

	* win32/common/libgstrtsp.def:
	* win32/common/libgsttag.def:
	Add new API functions to the dll exports

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2008-06-04  Michael Smith <msmith@songbirdnest.com>

	* gst/playback/gstplaybasebin.c:
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	  Disconnect signals from decodebins we created before we remove it
	  from playbin, to avoid crashes if the decodebin is eventually
	  disposed after the playbin itself (possible if the app takes a
	  reference on the decodebin).
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	  Fixes #536521.

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2008-06-04  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
	  (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
	  (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
	  (h264_video_type_find), (mpeg_video_stream_type_find),
	  (dv_type_find), (mmsh_type_find):
	  Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
	  copy caps for no good reason (this may be desirable to make it easier
	  to detect leaks, but then it should probably be done for all caps
	  in the typefinder somewhere).

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2008-06-04  Peter Kjellerstedt  <pkj@axis.com>

	* tests/check/Makefile.am:
	Do not try to run the check tests for subparse unless it has been
	built.

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	* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
	(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
	Do not try to run a test which requires vorbisenc unless we have
	actually built it.

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2008-06-04  Peter Kjellerstedt  <pkj@axis.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
	(gst_rtsp_connection_clear_auth_params),
	(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	Add a couple of missing argument guards.
	Add a way of setting the DSCP for an RTSP connection.
	Add an accessor method for the ip member of GstRTSPConnection as all
	members are supposed to be private.

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2008-06-04  Peter Kjellerstedt  <pkj@axis.com>

	* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
	Fixed accidental use of IPv4 options for all IPv6 addresses.

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2008-06-04  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/interfaces/mixertrack.h:
	  Document mixer track flags.

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	Patch by: Antoine Tremblay <hexa00 at gmail dot com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
	Don't set caps on the buffers that contain a copy of the buffer
	including the caps of them resulting in an always increasing refcount
	of the caps and insanely large caps. Instead include a buffer without
	caps in the new caps. Fixes bug #536475.

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2008-06-04  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
	Transform a given PAR to a range on the struct with the generic
	height/width instead of the struct with the possibly restricted
	height/width.

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	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
	Prefer the given format if it contains something stricter than [1,MAX]
	for height or width and only put a structure that requires rescaling
	as second. This makes it possible to use videoscale in pipelines where
	the source can actually produce the wanted height/width but usually
	selects a different one from the requested.

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2008-06-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	Based on patch by: John Millikin <jmillikin gmail com>

	* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
	  (gst_vorbis_tag_add_coverart):
	  Retrieve COVERART tags from vorbis comments (#512333)

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2008-06-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
	  Don't forget to add new enum value here too (should probably use
	  glib-mkenums here...).

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2008-06-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
	* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
	* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
	  (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
	  (gst_tag_image_data_to_image_buffer):
	  Add two utility functions to avoid code duplication (#512333):
	  API: add gst_tag_image_data_to_image_buffer()
	  API: add gst_tag_list_add_id3_image()
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2008-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* win32/common/libgstaudio.def:
	Add gst_audio_check_channel_positions() to the exported symbols.

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2008-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions):
	* gst-libs/gst/audio/multichannel.h:
	API: Make gst_audio_check_channel_positions() public.

	* tests/check/libs/audio.c: (GST_START_TEST):
	Add some simple checks for gst_audio_check_channel_positions().

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2008-06-02  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
	  minrange and maxrange are scaled according to the frequency
	  multiplier.

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2008-06-02  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* ext/pango/Makefile.am:
	* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
	  (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
	  Use gstvideo functions to calculate strides and plane offsets. Fixes
	  rendering issue ('ghost' images of the text on the chroma planes)
	  with widths or heights that are not multiples of 8 (#506659 and
	  probably also #485729).

	* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
	  (main):
	  Test with odd height/width too.

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	* gst/adder/gstadder.c: (gst_adder_query_duration),
	(gst_adder_query_latency):
	When using gst_element_iterate_pads() one has to unref every pad
	after usage.

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	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init):
	Add a gtk-doc chunk for the new properties to have a Since: indication.

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	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
	(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
	(gst_base_audio_src_change_state):
	Provide readable actual-buffer-time and actual-latency-time properties
	that reflect the configured ringbuffer values. Fixes #524724.
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	API: GstBaseAudioSrc:actual-latency-time
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	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
	(gst_basertppayload_change_state):
	Simply converting the running time into an RTP timestamp by scaling it
	based on the clock-rate is good enough for making an RTP timestamp. This
	has the added benefit that we can later on expose a property with the
	RTP timestamp of running time 0, as is needed for RTSP servers to
	generate the response of the PLAY request.

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2008-05-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c:
	(structure_has_fixed_channel_positions),
	(gst_audio_convert_transform_caps):
	Allow up to 11 positioned channels now that audioconvert can handle
	this but add no default positions for > 8 channels.
	
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Add some unit tests for the above change: Test conversion of
	11 positioned channels to stereo and the other way around, test
	conversion of 15 unpositioned channels in different ways.

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2008-05-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* win32/common/libgstaudio.def:
	Add gst_audio_clock_reset to the list of exported symbols.

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2008-05-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
	Remove wrong_channels_identification_header unit test as we now
	support 7 (and more channels).

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	* gst/audioconvert/gstchannelmix.c:
	(gst_channel_mix_fill_one_other):
	If mixing left or right to center (or the other way around) only take
	the complete value if we don't already have the original position in
	the source.

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	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions),
	(gst_audio_set_structure_channel_positions_list),
	(gst_audio_fixate_channel_positions):
	Allow rear center together with rear left/right and other previously
	conflicting channel positions. The reason why they weren't allowed
	was the channel mixing implementation in audioconvert.
	Also take this into account when fixing channel layouts.

	Allow setting channel positions for 1/2 channels when using
	gst_audio_set_structure_channel_position().

	* gst/audioconvert/gstchannelmix.c:
	(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_special), (gst_channel_mix_fill_matrix):
	Major rewrite of the channel mixing.

	We now allow previously	conflicting channel positions to appear
	together (rear center and rear left/right for example).
	Fixes bug #533817.

	Rework the way channels are mixed together to take more possible
	channel positions into account, properly mix from/to side channels
	and don't assume that either center, left&right or nothing of a
	specific position is available anymore.

	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Adjust unit tests with non-standard 1/2 channel layouts to the more
	correct new behaviour.

	Add a unit test for 5.1->Stereo downmixing.

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	* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
	Add sane defaults for the 7 and 8 channel layouts as those are
	undefined in the Vorbis spec. Use NONE channel layouts when decoding
	more than 8 channels instead of erroring out. Fixes bug #535356.

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	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* ext/theora/theoraparse.c:
	Add theoraparse to the docs and fix some docs.

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	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
	Fix EOS condition and track addition check, the track.end sector is
	included in the track. Fixes #533265.

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	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
	(gst_video_rate_flush_prev), (gst_video_rate_event),
	(gst_video_rate_chain):
	* gst/videorate/gstvideorate.h:
	React (more) to NEWSEGMENT
	Small adjustment in timestamp calculation to prevent mismatches
	Fixes #435633.

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	* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
	  Initialise error to NULL as we should.

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2008-05-28  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/adder/gstadder.c: (gst_adder_query_duration),
	(gst_adder_query_latency), (gst_adder_query):
	Implement latency query.

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2008-05-27  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/adder/gstadder.c: (gst_adder_query_duration):
	Correctly resync the iterator if gst_iterator_next() returns
	GST_ITERATOR_RESYNC.

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2008-05-27  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* win32/vs6/libgstpbutils.dsp:
	  Add pbutils-enumtypes.c to sources (#518037).

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	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
	* gst-libs/gst/audio/gstaudioclock.h:
	Add method to inform the clock that the time starts from 0 again. We use
	this info to calculate a clock offset so that the time we report in
	internal_time is monotonically increasing, as required by the clock base
	class. Fixes #521761.
	API: GstAudioClock::gst_audio_clock_reset()

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_change_state):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create), (gst_base_audio_src_change_state):
	Reset reported time when we (re)create the ringbuffer.

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2008-05-27  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* ext/alsa/gstalsamixertrack.c:
	  (gst_alsa_mixer_track_update_alsa_capabilities):
	  Make sure playback volumes aren't accidentally overwritten by
	  capture volumes if an alsa mixer track has both playback and
	  capture capabilities: we create two GstMixerTracks in that
	  case, so make sure we query only the alsa capabilities that
	  refer to the type of GstMixerTrack we created from the dual
	  capability alsa element. Should fix issues with Audigy2 sound
	  cards (#518082).

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2008-05-27  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* tests/check/pipelines/oggmux.c: (test_pipeline):
	  Don't use deprecated function.

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2008-05-27  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	(gst_decode_group_control_source_pad), (gst_decode_group_expose):
	Check for NULL cases and log them, creating ghostpads can, for example,
	fail when the pad returns wrong caps.

	* gst/playback/gstplaybin2.c: (perform_eos):
	When pushing out the EOS event, collect the return value and warn when
	something failed.

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2008-05-26  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	(gst_riff_create_video_template_caps):
	Add support for DVCPRO.

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2008-05-26  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
	  Change default scaling method from nearest-neighbour to bilinear.

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2008-05-26  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* tests/check/libs/video.c:
	  More checks.

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2008-05-25  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/subparse/gstsubparse.c: (parser_state_init),
	  (gst_sub_parse_format_autodetect), (handle_buffer):
	* gst/subparse/gstsubparse.h:
	* tests/check/elements/subparse.c: (test_tmplayer_style3b):
	  Limit duration to a maximum of five seconds for tmplayer format where
	  we can guess the duration only from the timestamp of the next line of
	  text. We don't want to show a text for eternities just because nothing
	  else is being said for a while.

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2008-05-23  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_push_full),
	(gst_base_rtp_depayload_change_state):
	Check sequence numbers, mark input buffers with a discont flag for the
	subclass when we detected a gap, drop duplicate buffers. We do this
	because one can use the element without a jitterbuffer in front and we
	don't want to feed the subclasses invalid or reordered data.
	Do an error when the subclass did not provide a process function instead
	of crashing.
	Some other small cleanups.

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2008-05-22  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
	  May just as well use the precalculated uvstride here.

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2008-05-22  Jan Schmidt  <jan.schmidt@sun.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-overrides.txt:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* docs/plugins/gst-plugins-base-plugins.interfaces:
	* docs/plugins/gst-plugins-base-plugins.prerequisites:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gio.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playback.xml:
	* docs/plugins/inspect/plugin-queue2.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-uridecodebin.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* ext/cdparanoia/gstcdparanoiasrc.c:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggdemux.h:
	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggmux.h:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/gdp/gstgdpdepay.h:
	* gst/gdp/gstgdppay.h:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gsturidecodebin.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gstmultifdsink.h:
	* gst/tcp/gsttcp.h:

	Add some documentation comments, and some new headers to be scanned.
	Rename some internal enum declarations (audioconvert's DitherType and
	NoiseShapingType, GstUnitType from the TCP elements) to match the
	documented GObject type names so that the docs pick them up.
	Name the playbin2 docs markups properly so they get picked up. They'll
	need renaming back when/if playbin2 becomes playbin.

	100% symbol coverage for the plugin docs, booya.

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2008-05-22  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
	Fix generation of NV12/NV21 frames. Fixes bug #532454.

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2008-05-22  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* gst/playback/gstdecodebin.c: (remove_fakesink):
	Lock the fakesink before setting the state to NULL and removing it from
	the bin so that a concurrent state change cannot interfere. 
	Fixes #534331.

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2008-05-21  Felipe Contreras  <felipe.contreras@gmail.com>

	* docs/Makefile.am:
	Fix installing plugin documentation when gtk-doc is disabled.

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2008-05-21  Felipe Contreras  <felipe.contreras@gmail.com>

	* gst-libs/gst/rtsp/Makefile.am:
	Distribute, don't install md5.h

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2008-05-21  Julien Moutte  <julien@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
	instead of SOL_IP, works on more platforms.
	* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
	arguments.

Wim Taymans's avatar
Wim Taymans committed
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2008-05-21  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/vorbis/vorbisdec.c:
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
	Some debug and comment fixes.

	* tests/examples/dynamic/addstream.c: (main):
	Fix , to ;

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2008-05-21  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
	* gst/playback/decodetest.c: (new_decoded_pad_cb):
	* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
	(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
	(cleanup_decodebin):
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
	(connect_element), (gst_decode_group_control_demuxer_pad):
	* gst/playback/gstplaybasebin.c: (queue_remove_probe),
	(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
	(mute_group_type):
	* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
	(gst_play_bin_set_property), (handoff), (gen_video_element),
	(gen_text_element), (gen_audio_element), (gen_vis_element),
	(remove_sinks), (add_sink), (setup_sinks):
	* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
	* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
	(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
	(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
	(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
	(gen_video_chain), (gen_text_chain), (gen_audio_chain),
	(gen_vis_chain), (gst_play_sink_reconfigure),
	(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
	(gst_play_sink_request_pad):
	* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
	* gst/playback/test.c: (gen_video_element), (gen_audio_element),
	(cb_newpad):
	* gst/playback/test6.c: (new_decoded_pad_cb):
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	* tests/check/elements/audiorate.c: (test_injector_chain),
	(do_perfect_stream_test):
	* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
	* tests/check/elements/gdpdepay.c: (GST_START_TEST):
	* tests/check/elements/gnomevfssink.c:
	* tests/check/elements/textoverlay.c:
	(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
	* tests/check/elements/videotestsrc.c: (GST_START_TEST):
	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (test_pipeline):
	* tests/check/pipelines/streamheader.c: (GST_START_TEST):
	* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
	* tests/examples/seek/scrubby.c: (make_wav_pipeline):
	* tests/examples/seek/seek.c: (make_mod_pipeline),
	(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
	(make_sid_pipeline), (make_parse_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),
	(update_fill), (msg_buffering):
	Don't use bad gst_element_get_pad().

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2008-05-21  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-media.c:
	  Fix wrong method name in docs. Fix calculation of strf fields for
	  broken mulaw/alaw.
  
	* gst-libs/gst/riff/riff-read.c:
	  Whitespace fix and removing double ';'.

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2008-05-21  Wim Taymans  <wim.taymans@collabora.co.uk>

	* docs/design/part-playbin2.txt:
	Add some leftover doc.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
	Fix copy & paste error in last commit.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
	Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
	other channel positions when source has SIDE channels and dest doesn't
	or the other way around.

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2008-05-21  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Henrik Eriksson <henriken at axis dot com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
	(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
	(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
	(gst_multi_fd_sink_get_property):
	* gst/tcp/gstmultifdsink.h:
	Add support for DSCP QOS. Fixes #469933.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Add another test that checks if conversion between standard 1 and 2
	channel layouts with and without positions set is working.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions):
	Allow non-standard 2 channel layouts.
	
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Add some tests for converting and remapping non-standard 1 and 2
	channel layouts.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstchannelmix.c:
	(gst_channel_mix_fill_normalize):
	Prevent division by zero if the channel mix matrix contains only
	zeroes.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Antoine Tremblay <hexa00 at gmail dot com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
	Close a buffer memory leak. Fixes bug #534071.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/rtsp/gstrtsptransport.h:
	Make the GstRTSPTransport struct members public as there are no
	setters/getters and it's supposed to be changed directly.
	Fixes bug #533087.

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2008-05-21  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	Adder also doesn't support audio/x-raw-int with width!=depth so don't
	claim this on the pad template caps.

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2008-05-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_sync_latency):
	We can only use our optimal calibration if we prerolled before the
	latency expired.

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	* configure.ac:
	  Require core CVS for GstBaseSrc buffer caps setting magic.

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2008-05-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_fixate_channels):
	Fix logic in last commit.

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2008-05-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_fixate_channels):
	Passthrough the channel positions if the number of output channels is
	the same as the number of input channels, the input had a channel
	layout and downstream requests no special one. We did this already for
	> 2 channels but now it's also done for 1 channel. Fixes bug #533617.

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2008-05-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
	(gst_gnome_vfs_src_finalize),
	(gst_gnome_vfs_src_received_headers_callback),
	(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
	* ext/gnomevfs/gstgnomevfssrc.h:
	Set the ICY caps on the srcpad from where they get picked up by the base
	class now and set on the outgoing buffers.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
	BaseSrc now sets the caps on outgoing buffers automatically.

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2008-05-20  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_resample_slaving),
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
	(gst_base_audio_sink_async_play),
	(gst_base_audio_sink_change_state):
	Change the way in which the ringbuffer is started when dealing with a
	slaved clock and latency. We now sync to the clock until we reach
	upstream latency before starting the ringbuffer. This has the effect
	that we can accurately align the master and slave clocks and let the
	rate correction code take care of the initial drift or rounding errors
	instead of leaving them uncorrected with the old approach.

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2008-05-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_fixate_channels):
	Correctly set the default channel positions when converting to 8
	channels.

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2008-05-19  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* configure.ac:
	  Error out if we don't have the required version of core.

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2008-05-19  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
	  Use data scan helper in aac typefinder and stop scanning
	  for headers when we've found a type. Also fix potential invalid
	  memory access when calculating the frame length.

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2008-05-19  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
	  (mpeg_sys_is_valid_pack):
	  Don't modify scan context when we return FALSE in ensure_data, so
	  it's possible to continue scanning, and we don't end up with a NULL
	  data pointer and a positive size, which might bite us the next time
	  we're called. Small constification.

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2008-05-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/adder/gstadder.c:
	Adder doesn't support 24 bit samples so don't claim it supports them
	in the pad template caps.

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2008-05-14  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain):
	Validate the RTP packet before further processing it. It's just too
	dangerous to accept random packets and people are not forced to use a
	jitterbuffer or session manager to filter out the bad packets.

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	(gst_rtp_buffer_set_extension_data),
	(gst_rtp_buffer_get_payload_subbuffer):
	Small cleanups.
	When setting extension data in a buffer that is too small, we fail and
	we should not set the extension bit.
	Change GST_WARNINGS into g_warning because they really are
	programming errors.

	* tests/check/libs/rtp.c: (GST_START_TEST):
	Catch the g_warnings now in the unit tests and that fact that failing to
	set extension data left the extension bit untouched.

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2008-05-14  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
	  Revert previous change which made basetransform handle buffer_alloc
	  and which breaks things badly in the non-passthrough case since it
	  returned buffers with a different (ie. sometimes smaller) size than
	  the size requested.

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2008-05-14  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Bernard B <b-gnome at largestprime dot net>

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
	Fix seqnum compare function for bordercase values and fix the docs
	again. Fixes #533075.

	* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
	Add a testcase for seqnum compare function.

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2008-05-14  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/adder/gstadder.c: (gst_adder_setcaps),
	(gst_adder_class_init):
	Correctly declare the supported endianness on the pad templates
	and check for correct endianness in the set caps function. Adder
	only supports native endianness.
	Also use gst_element_class_set_details_simple().

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2008-05-14  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c:
	  Better debug logging in port value handling. Merging separate port
	  value loops into one.

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2008-05-13  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Hannes Bistry <hannesb at gmx dot de>

	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
	* gst/tcp/gsttcpserversink.c:
	(gst_tcp_server_sink_handle_server_read),
	(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
	Fix regression in clientsrc because we did not add the fd to the poll
	set anymore. Fixes #532364.
	Do some cleanups here and there.

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2008-05-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
	* gst/playback/gstplay-marshal.list:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
	Use correct marshallers. GstCaps are a boxed type and no GObject
	subclass.

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2008-05-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* win32/common/libgstrtsp.def:
	Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
	symbols.

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2008-05-13  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* tests/check/elements/audioresample.c:
	(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
	(live_switch_push), (GST_START_TEST):
	Add unit test for the latest basetransform negotiation changes.
	See bug #526768.

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2008-05-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
	Fix nv12<->nv21 conversion if stride is larger than width.

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	Patch by: Jan Gerber <j at oil21 dot org>
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	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
	(gst_ogg_pad_parse_skeleton_fisbone):
	* ext/ogg/gstoggdemux.h:
	Parse presentation time from skeleton streams and use it as offset
	for the timestamps. Fixes bug #530068.

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2008-05-12  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
	Revert previous patch that attempted to more accurately calculate the
	initial offset between master and slave clock. The best thing we can do
	in general is take the time of both clocks as the diff since we don't
	know when the actual preroll happened.

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2008-05-11  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/pbutils/install-plugins.c:
	  Fix docs: type and missing word.

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2008-05-10  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
	  Don't do lots of 4-byte peeks, but use the 'new' data scan helper
	  for this instead; don't check if we've found enough markers after
	  each and every step, it's enough to do that only if we've actually
	  found a new marker.
	  Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.

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2008-05-10  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c:
	  (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
	  (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
	  (mpeg_video_stream_type_find):
	  Move scan helper thingy to the beginning of the file so we can use
	  it in other typefind functions. Rename it to something more
	  generic. Also improve handling of things towards the end of the
	  typefind data: peek as much as we can if we know the size of the
	  data, rather than just min_size.

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2008-05-09  Jan Schmidt  <jan.schmidt@sun.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/interfaces/colorbalance.c:
	* gst-libs/gst/interfaces/colorbalance.h:
	* gst-libs/gst/interfaces/colorbalancechannel.c:
	* gst-libs/gst/interfaces/colorbalancechannel.h:
	* gst-libs/gst/interfaces/tuner.c:
	* gst-libs/gst/interfaces/tunerchannel.c:
	* gst-libs/gst/interfaces/tunerchannel.h:
	* gst-libs/gst/interfaces/tunernorm.c:
	* gst-libs/gst/interfaces/tunernorm.h:
	* gst-libs/gst/video/video.c:
	* gst-libs/gst/video/video.h:
	Document the GstTuner and GstColorBalance interfaces, and some
	other random API functions that needed it. 70% symbol coverage, woo.

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2008-05-09  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
	Choose to allocate one less segment but require one additional segment
	as latency. 

	* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
	No need to increment the number of segments in the source.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_get_time), (clock_convert_external),
	(gst_base_audio_sink_resample_slaving),
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_async_play):
	Remove adding latency when returning the internal time while subtracting
	it again when we use the value a little later.
	When calculating the end timestamp, we are making a rounding error
	with the current algorithm. Ensure that we don't accumulate these
	rounding errors when aligning samples by not resampling at all if we
	don't need to. Fixes #419351.
	Make the initial calibration of the clock slaving a little more
	predictable and accurate. Also handle the case where we don't do
	clock slaving.

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2008-05-09  Sebastian Dröge  <slomo@circular-chaos.org>

	Based on a patch by:
	  Björn Benderius <bjoern dot benderius at axis dot com>

	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	(gst_ffmpegcsp_avpicture_fill):
	* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
	* gst/ffmpegcolorspace/imgconvert_template.h:
	Add conversions from/to NV12 and NV21 and conversions between those
	two formats. Fixes bug #532166.

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2008-05-08  Edward Hervey  <edward.hervey@collabora.co.uk>

	* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
	Abort the h264 typefinding as soon as _peek() doesn't return anything,
	which happens for example with files smaller than 128kb.

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2008-05-08  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Wouter Cloetens <zombie at e2big dot org>

	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_create), (md5_digest_to_hex_string),
	(auth_digest_compute_hex_urp), (auth_digest_compute_response),
	(add_auth_header), (gst_rtsp_connection_free),
	(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
	(gst_rtsp_connection_set_auth_param),
	(gst_rtsp_connection_clear_auth_params):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	Add Digest authorization support for RTSP connections. See #532065.

	* gst-libs/gst/rtsp/md5.c:
	* gst-libs/gst/rtsp/md5.h:
	Yeap, another md5 implementation until we can depend on a glib that has
	support for it.

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2008-05-08  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
	Let audioresample use the buffer allocation of basetransform instead
	of it's own stuff.

	* tests/check/elements/audioresample.c: (alloc_only_48000),
	(GST_START_TEST), (audioresample_suite):
	Add unit test for the recent basetransform bugfix, where upstream
	changes caps to something that can't be passed through anymore.

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2008-05-07  Ole André Vadla Ravnås  <ole.andre.ravnas at tandberg com>

	* win32/common/config.h.in:
	  Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
	  use the real thing than having "???" unconditionally.

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2008-05-07  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_query):
	Report the latency with the new seglatency parameter.

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_acquire):
	* gst-libs/gst/audio/gstringbuffer.h:
	Add new field to the ringbufferspec to specify the expected latency
	between the underlying device read/write pointer, this is needed
	when writing sinks that sit a little closer to the hardware.
	Add some more docs for other fields.

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2008-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/volume/gstvolume.c: (volume_transform_ip):
	Return NOT_NEGOTIATED if we didn't set a process function yet for some
	reason instead of crashing later. Might fix bug #509125.

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2008-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_parse_caps),
	(structure_has_fixed_channel_positions),
	(gst_audio_convert_transform_caps):
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
	Add support for more than 8 channels and NONE channel layouts. For
	more than 8 channels no channel conversion is supported yet, only
	format conversions are supported. Fixes bug #398033.

	* tests/check/elements/audioconvert.c: (verify_convert),
	(GST_START_TEST), (audioconvert_suite):
	Add some unit tests by Tim for checking the NONE channel layouts
	and more than 8 channels and add some more unit tests for channel
	conversions.

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2008-05-06  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst/playback/gstdecodebin2.c: (connect_pad):
	When autoplugging fails, set the element back to NULL before
	unreffing it.

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2008-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* win32/common/libgstaudio.def:
	Add gst_base_audio_src_[sg]et_slave_method() to the exported
	symbols.

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2008-05-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/subparse/samiparse.c: (handle_start_sync),
	(end_sami_element), (characters_sami):
	Remove trailing, leading and double whitespaces.
	Correctly timestamp buffers and output the last buffer too.

	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	Add a simple unit test for SAMI parsing.

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2008-05-05  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/subparse/samiparse.c: (handle_start_sync),
	(start_sami_element), (end_sami_element), (characters_sami),
	(sami_context_reset):
	Only output characters inside the "sync" elements. There could be
	other elements like "style" that have some content but should
	not be printed. Fixes bug #467911.

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2008-05-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/playback/gstplaybasebin.c: (set_audio_mute),
	(set_active_source):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(playbin_set_audio_mute):
	Allow setting -1 as current-audio to mute the current audio stream,
	similar to what is done for subtitles. Fixes bug #342294.

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2008-05-05  Edward Hervey  <edward.hervey at collabora co uk>

	* gst-libs/gst/pbutils/descriptions.c: (formats): 
	It's SorensOn and not SorensEn.

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2008-05-04  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst-libs/gst/pbutils/descriptions.c: (formats):
	  Fix description of video/x-flash-video.

Sebastian Dröge's avatar
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2008-05-04  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
	* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
	Remove some unused code.

	* gst/audioconvert/gstaudioquantize.c:
	(gst_audio_quantize_free_noise_shaping):
	Don't return before freeing the noise shaping history.

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2008-05-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* tests/check/elements/subparse.c: (do_test),
	  (test_tmplayer_style3b), (subparse_suite):
	  Add unit test for the tmplayer variant from bug #530962.

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2008-05-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/subparse/gstsubparse.c: (handle_buffer),
	  (gst_sub_parse_sink_event):
	* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
	  (tmplayer_parse_line):
	  Fix parsing of tmplayer subtitle variant where every single line contains
	  text and there isn't an empty line after each line to determine the
	  duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
	  making sure that we push out the last line of text without a duration if
	  there's still text left in the buffer at the end.

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2008-05-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/subparse/gstsubparse.c: (feed_textbuf):
	  Fix detection of discontinuities based on the buffer offset (doesn't work
	  so well if no buffer offset is set) and also check for the DISCONT buffer
	  flag. This keeps the parser state from being reset after each buffer in
	  the unit test.

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2008-05-03  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
	  Further fine-tuning: don't absolutely require sequence or GOP headers
	  (as introduced in the previous commit), but adjust the typefind
	  probabilities returned accordingly if we don't see them. Also make sure
	  picture header and first slice are somewhat close to each other (which
	  is not perfect but still better than requiring a fixed offset or having
	  no limit at all).

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2008-05-02  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
	(gst_basertppayload_sink_setcaps),
	(gst_basertppayload_sink_getcaps):
	Rename the setcaps/getcaps function internally to make it clear that
	they are called for the sink pad.

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2008-05-02  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_class_init),
	(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
	(gst_base_rtp_depayload_packet_lost),
	(gst_base_rtp_depayload_set_gst_timestamp):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Catch packet-lost events from the jitterbuffer and convert them into a
	vmethod call (lost-packet) so that depayloaders can do something smart.
	Also add a default packet-lost function that sends out a segment update
	to the decoders.

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	* gst/playback/test4.c:
	* gst/playback/test5.c:
	* gst/playback/test6.c:
	* gst/playback/test7.c:
	  Also include config.h when relying on defines from it. Fixes the
	  build. Its been a please to serve :)

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2008-05-02  Thijs Vermeir <thijsvermeir@gmail.com>

	* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
	(paint_setup_NV21), (paint_hline_NV12_NV21):
	Add support for NV12 and NV21 in videotestsrc

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2008-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/videoscale/gstvideoscale.c:
	* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
	* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
	(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
	(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
	(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
	(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
	(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
	(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
	(vs_image_scale_linear_RGB555):
	Support 1x1 images as input and output as for example the BBC HQ new
	streams have 1x1 GIFs in the playlists for some reason.

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2008-05-01  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
	  (try_to_link_1):
	  If we can't activate one of the decoders we plugged in (such as,
	  say, musepackdec) for some reason (it might not support push mode,
	  for example), remove any pad probes that close_pad_link() might
	  have set up. This makes sure we later don't try to remove a probe
	  for a pad that doesn't exist any longer, and avoids nast warnings
	  and probably other things too.

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2008-04-30  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c:
	  (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
	  (plugin_init):
	  Rework mpeg video stream typefinding a bit more: make sure sequence,
	  GOP, picture and slice headers appear in the order they should and
	  that we've in fact at least had one of each; fix picture header
	  detection; decouple picture and slice header check - don't assume
	  they're at a fixed offset, there may be extra data in between. Also,
	  announce varying degrees of probability depending on what we found
	  exactly (multiple pictures, at least one picture, just sequence and
	  GOP headers). Finally, in _ensure_data(), take into account that we
	  might be typefinding smaller amounts of data, such as the first
	  buffer of a stream, so fall back to the minimum size needed as long
	  as that's available, instead of erroring out if there's less than
	  2kB of data. Fixes #526173. Conveniently also doesn't recognise the
	  fuzzed file from #399342 as valid.

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2008-04-30  Michael Smith <msmith@songbirdnest.com>

	* ext/theora/theoradec.c:
	  Cool kids don't divide by zero.
	  Treat PAR of x:0 as 1:1.
	  Fixes #530719.

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2008-04-30  Tim-Philipp Müller  <tim.muller at collabora co uk>

	* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
	  (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
	  (mpeg_video_stream_type_find):
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	  Refactor a bit: use context structure to track parsing offset and
	  size of available data and make the code a bit clearer. Fixes bad
	  memory access in #356937.
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2008-04-28  Michael Smith <msmith@songbirdnest.com>

	* gst/playback/test4.c:
	* gst/playback/test5.c:
	* gst/playback/test6.c:
	* gst/tcp/gstmultifdsink.c:
	  Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
	  is defined.

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2008-04-28  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/audio/gstbaseaudiosink.h:
	Clarify some docs.

	* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
	(gst_base_audio_src_class_init), (gst_base_audio_src_init),
	(gst_base_audio_src_set_slave_method),
	(gst_base_audio_src_get_slave_method),
	(gst_base_audio_src_set_property),
	(gst_base_audio_src_get_property), (gst_base_audio_src_create):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	Add property and methods for selecting the clock slave method in the
	source, like in the sink.
	We only implement "none" and "re-timestamp" for now.
	API: gst_base_audio_src_set_slave_method()
	API: gst_base_audio_src_get_slave_method()

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2008-04-25  Wim Taymans  <wim.taymans@collabora.co.uk>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	Fix the docs about the seqnum compare function, it returns a difference.

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2008-04-24  Edward Hervey  <edward.hervey@collabora.co.uk>

	* ext/alsa/gstalsadeviceprobe.c:
	(gst_alsa_get_device_list): Don't return before freeing up
	the allocated structures.

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