ChangeLog 1.13 MB
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2007-04-12  Michael Smith  <msmith@fluendo.com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Patch by: René Stadler  <mail at renestadler de>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-03-20  Michael Smith  <msmith@fluendo.com>

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

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2007-03-19  Wim Taymans  <wim@fluendo.com>

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

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

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

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

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

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

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

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

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

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

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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	Add min-ptime property to RTP base audio payloader. Patch by
	olivier.crete@collabora.co.uk.
	Fixes #415001

	Indentation/whitespace/documentation fixes.

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2007-03-14  Julien MOUTTE  <julien@moutte.net>

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

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

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update translations from translation project.

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

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

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

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

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

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

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

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

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

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

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

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	Patch by: Kamil Pawlowski  <kamilpe gmail com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* configure.ac:
	Back to CVS

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=== release 0.10.12 ===

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

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

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

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

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2007-03-06  Wim Taymans  <wim@fluendo.com>

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

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

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

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2007-03-05  Sebastian Dröge  <slomo@circular-chaos.org>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* configure.ac:
	  Convert to new AG_GST style.

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

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

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2007-02-28  Jan Schmidt  <thaytan@mad.scientist.com>

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

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

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

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

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

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

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

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

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

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

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2007-02-26  Wim Taymans  <wim@fluendo.com>

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

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

	Patch by: René Stadler <mail at renestadler de>

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

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

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

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

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

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

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

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2007-02-22  Tim-Philipp Müller  <tim at centricular dot net>

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

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

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

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

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2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

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

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2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

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

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2007-02-21  Stefan Kost  <ensonic@users.sf.net>

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

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2007-02-21  Stefan Kost  <ensonic@users.sf.net>

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

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2007-02-20  Edward Hervey  <edward@fluendo.com>

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

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2007-02-20  Edward Hervey  <edward@fluendo.com>

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

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Yves Lefebvre  <ivanohe abacom com>

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

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2007-02-15  Stefan Kost  <ensonic@users.sf.net>

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

Wim Taymans's avatar
Wim Taymans committed
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2007-02-15  Wim Taymans  <wim@fluendo.com>

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

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