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2007-10-22  Edgard Lima  <edgard.lima@indt.org.br>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too
	  early. It is temporary until we find something better.

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved):
	Fix race when pausing a RTSP stream in interleaved.
	Fixes #475784.

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Peter Kjellerstedt <pkj at axis com>

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize):
	Use correct unref function for buffers. #488844.

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2007-10-19  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* tests/check/elements/avimux.c:
	  Add some debug and sync tests with the fix.	  

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	When the socket is used by the app for other purposes, don't generate an
	error if there is activaty on the socket that is not data related.
	Fixes #487488.

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	(gst_v4l2src_grab_frame):
	Add some more debug info. Generate an error when we run out of buffers
	for some reason. See #480557.

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Anders Skargren <anders dot skargren at axis dot com>

	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Set marker bit correctly.

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2007-10-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	Use allowed name for the GstStructure.

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2007-10-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	* gst/autodetect/gstautoaudiosink.c:
	  Use new gst_bus_pop_filtered().

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2007-10-13  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  When probing the formats and sizes a camera supports, make
	  sure the best ones (highest resolution, prefered format)
	  end up at the beginning of the probed caps and the less
	  desirable ones at the end.  This is important because the
	  order within the caps matters for things like fixation and
	  negotiation, ie. what format is chosen in the end.
	  With recent kernels, the current probing code will end up
	  querying the supported sizes from lowest resolution to
	  highest resolution, adding them to the probed caps in that
	  order, resulting to v4l2src fixating to the lowest possible
	  resolution if downstream does not express a size preference.
	  Also make up a somewhat random ranking of prefered output
	  formats for the same reason. Fixes #485828.
	
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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jason Kivlighn  <jkivlighn gmail com>

	* gst/id3demux/id3v2frames.c:
	  Extract license/copyright URIs from ID3v2 WCOP frames
	  (Fixes #447000).

	* tests/check/elements/id3demux.c:
	* tests/files/Makefile.am:
	* tests/files/id3-447000-wcop.tag:
	  Add simple unit test.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Add support for license/copyright URI tags (ID3v2 WCOP frame).
	  Prerequisite for #447000.

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2007-10-08  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
	Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
	a GstClockTime.

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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	More seeking fixes, mostly passing around the new playback segment in
	order to configure it properly.
	Also reset base_time of udp sources when setting them back to PLAYING as
	a temporary hack until core supports seek in live sources properly.

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2007-10-08  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpmp4adepay.c:
	Fix caps as to not confuse autopluggers.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/gstid3demux.h:
	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	  Port ID3 tag demuxer over to the new GstTagDemux in -base
	  (now would be a good time to test re-importing your music
	  collection).

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/Makefile.am:
	* gst/apetag/gstapedemux.c:
	* gst/apetag/gstapedemux.h:
	* gst/apetag/gsttagdemux.c:
	* gst/apetag/gsttagdemux.h:
	  Port APE tag demuxer over to the new GstTagDemux in -base.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_internal_src_query),
	(gst_rtspsrc_handle_src_query), (new_session_pad),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_loop_send_cmd):
	Improve flushing behaviour.
	Set state of the udp sources to PAUSE/PLAYING correctly.
	Handle events and queries for UDP and TCP transport now.

Stefan Kost's avatar
Stefan Kost committed
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2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	  Add log category.

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2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Timo Hotti <Timo.Hotti@sysopendigia.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c:
	  Add unit tests for payloaders/depayloaders.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  Also save codec data for audio streams. Fixes #482495.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  Fix "Index entry has invalid stream nr 1".
	  Add support for muxing aac - work in progress (see #482495).

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
	(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	* gst/rtsp/gstrtspsrc.h:
	Parse bandwidth modifiers, they are not yet configured in the session
	manager because we don't have an API for that yet.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	Use shiny new function in -base to get the default clock-rate.
	Update some docs.

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2007-09-29  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add files to win32 manifest.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.

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2007-09-28  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	In TCP mode, only timestamp the first buffer. TCP is not real time and
	it does not make sense to try to skew compensate, also some servers send
	the first batch of data in a burst.

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2007-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c:
	  Fix setting the discont flag on the first buffer
	  pushed downstream for formats with private codec
	  data that needs to be deserialised into buffers
	  (such as vorbis and FLAC when in a matroska container).

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2007-09-27  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
	(gst_rtp_mp4v_pay_handle_buffer):
	* gst/rtp/gstrtpmp4vpay.h:
	Free the config string. Fixes #480707.
	Clean up the timestamp code a little.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	Set timestamps on RTP buffers in interleaved mode.
	Mark first buffers with a DISCONT.
	Remove flush hack now that sync for live sources has been figured out.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Update documentation.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
	(gst_rtp_xqt_depay_change_state):
	* gst/qtdemux/gstrtpxqtdepay.h:
	Fail if we don't know the quicktime format.

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2007-09-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  Save the flow return from the last gst_pad_push() and
	  make sure we pass the right flow return value upstream
	  in the case of failure; minor clean-ups.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* gst/apetag/gstapedemux.c:
	  Add support for the new GST_TAG_COMPOSER (#459809).

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/alaw.c:
	* gst/law/mulaw-conversion.h:
	  Compulsive clean-ups: use boilerplate macros, add debug
	  categories, fix up things to conform to symbol nomenklatura,
	  etc.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	  Use static tables for A-Law decoding and encoding; this makes
	  A-Law decoding and encoding less CPU-intensive, but increases
	  the binary size a bit. Leaving old code around for now,
	  selectable by a define in the code. Fixes #435435.

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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
	AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros
	in configure.ac.

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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: <j at bootlab dot org>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add fourccs for MPEG2 HDV streams. Fixes #479960.

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2007-09-23  Stefan Kost  <ensonic@users.sf.net>

	* sys/oss/gstosshelper.c:
	  Use GST_WARNING instead of a g_critical. This situation is not caused
	  by the application.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/LINGUAS:
	* po/nl.po:
	  Updated translations.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mikel Olasagasti <hey_neken@mundurat.net>

	* po/eu.po:
	  Added Basque translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Abel Cheung <abelcheung@gmail.com>

	* po/zh_HK.po:
	* po/zh_TW.po:
	  Added Chinese (traditional and Hong Kong) translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jakub Bogusz <qboosh@pld-linux.org>

	* po/pl.po:
	  Added Polish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Ilkka Tuohela <hile@iki.fi>

	* po/fi.po:
	  Added Finnish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jorge González González <aloriel@gmail.com>

	* po/es.po:
	  Added Spanish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mogens Jaeger <mogens@jaeger.tf>

	* po/da.po:
	  Added Danish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Funda Wang <fundawang@linux.net.cn>

	* po/zh_CN.po:
	  Added Chinese (simplified) translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Alexander Shopov <ash@contact.bg>

	* po/bg.po:
	  Added Bulgarian translation.

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2007-09-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
	Set outgoing packet duration because we can. Fixes #478244 some more.

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2007-09-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/cairo/gsttextoverlay.c:
	  Add info about static leak.
	
	* tests/check/Makefile.am:
	* tests/check/generic/states.c:
	  Improved state change unit test.

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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/.cvsignore:
	* tests/check/.cvsignore:
	  Ignore registries in any format.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
	Removed some unused code.

	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
	* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
	(gst_rtp_theora_pay_flush_packet):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
	Try to preserve the incomming buffer duration on the outgoing
	packets. Fixes #478244.

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2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	  Work around compiler warnings with g++-4.2 when assigning a
	  string constant to a gchar * (partially fixes #478092).

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2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We require core CVS now for gst_base_src_set_do_timestamp().

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2007-09-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message):
	Fix compiler warnings shown with Forte.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_dup_printf):
	Give meaningfull error when all streams failed to configure for some
	reason.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/README:
	Update README with the design for synchronisation rules of RTP on
	sender and receiver.

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2007-09-14  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
	(gst_wavparse_chain):
	Don't push EOS from the chain function, the element
	driving the pipeline is responsible for this. The bug
	this was meant to fix seems to be queue not forwarding
	EOS in all cases (see #476514).

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2007-09-13  Wim Taymans  <wim.taymans@gmail.com>

	* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	Use basetransform segment so that it is correctly managed on flushes and
	start/stop.
	Report message timestamp as stream time, which is what an application
	can understand.

Sebastian Dröge's avatar
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2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gsttaglibmux.c:
	* tests/check/elements/apev2mux.c:
	Update my mail address.

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2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
	(gst_wavparse_loop), (gst_wavparse_chain):
	Add EOS logic for the push-based mode too. Fixes #476514.

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2007-09-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
	* gst/law/alaw-encode.h:
	* gst/law/mulaw-encode.c: (gst_mulawenc_init),
	(gst_mulawenc_chain):
	* gst/law/mulaw-encode.h:
	Fix law encoder timestamps.

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2007-09-12  Stefan Kost  <ensonic@users.sf.net>

	* ext/gconf/gstgconfaudiosink.c:
	  Fix warning when building without debug.

	* sys/oss/gstossmixertrack.c:
	  Use const like in alsamixertrack.c (fixes warnings).

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2007-09-11  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Fix framerate detection code some more.
	Handle the case where there is a weird step in the stepwise framerates.
	Don't overwrite the min interval with the framerate, use a temp variable
	instead.
	Use max in the Continuous framerate intervals instead of step, which is
	1 according to the docs. Fixes #475424.

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2007-09-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
	Make udpsrc timestamp outgoing buffers based on when they were received.
	Also make it output a segment in time.

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2007-09-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Plug a little leak. Little code cleanups.

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

	* configure.ac:
	  Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
	  flac versions, 's good for cross-compilation karma.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>

	* gst/rtp/gstrtph263pay.c:
	  Fix up header structure so that compilers don't add padding
	  between the structure fields, since that would lead to us
	  sending RTP packets with broken headers (as is currently the
	  case when compiling with MSVC). Also see similar fixes in
	  libgstrtp in gst-plugins-base. (#474616; #471194)

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2007-09-07  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Don't overwrite our GValue with 0 but instead use the previously
	computed value. Fixes #471823 some more.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  No tabs in this file please, or gtk-doc will end up documenting
	  rather absurd class hierarchies.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	  If the new kid element fails to change state for some reason
	  (e.g. esdsink not being able to connect to the sound server),
	  forward the error message it posted on the bus instead of just
	  posting a generic 'Internal state change error: please file a
	  bug' error message. Fixes #471364.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  Don't assume tags are encoded as UTF-8 (#473670).

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	  Implement LATENCY queries in the crudest way possible so I don't
	  have to use sync=false any longer when testing with videosinks.

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Tim-Philipp Müller committed
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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix build.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Add some more debugging in the framerate function.
	Iterate stepwise framerate up to and _including_ the max and if nothing
	was added to the list, add a dummy 0/1 to 100/1 framerate so that we
584
	don't end up with an empty list. Fixes #471823
585

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_set_clients_string),
	(gst_multiudpsink_get_clients_string),
	(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
	(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
	(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
	(gst_multiudpsink_clear):
	Add property do configure destination address/port pairs
	API:GstMultiUDPSink::clients

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/Makefile.am:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sdp:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H264-PCMA.sdp:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	Added some RTP example scripts for sending and receiving RTP streams.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
	(gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
	Restructure the setcaps function so that we can also compute the
	expected GStreamer output size of the video frames.
	Set frame_byte_size correctly so that read-based devices have a chance
	of working correctly.
	When grabbing a frame, discard frames that are not of the expected size.
	Some cameras don't output the right framesize for the first buffer.
	Try only a couple of times to get a valid frame, else error out.

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	(gst_v4l2_fill_lists), (gst_v4l2_get_input):
	Add some more debug info when scanning the device.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
	(gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
	(gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
	(gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
	Add some more debug info when dequeing a frame.

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2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  More code cleanups. Add some more comment and improve debugs logs.

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2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  Implement seek-query. Refactor duration calculations. Appropriate use
	  of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
	  out of loops.

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2007-09-03  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Implement seek-query.

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2007-08-29  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_dup_printf):
	Use new basesink async property to make sparse RTCP packet not wait for
	preroll.

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2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audiofx/Makefile.am:
	Dist the right file.

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2007-08-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
	(gst_rtspsrc_get_float), (gst_rtspsrc_play):
	Make sure we generate and parse floating point values in the POSIX
	locale instead of the current locale. 

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2007-08-22  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Fix method detection again.
	Keep track of when we must send a Range header.
	Use segment values for Range, Speed and Scale headers.
	Parse Speed and Scale headers to update the segment values.

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2007-08-22  Stefan Kost  <ensonic@users.sf.net>

	patch by: Mark Nauwelaerts <manauw@skynet.be>

	* sys/v4l2/v4l2src_calls.c:
	  Handle optional v4l2 ioctls gracefully.

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2007-08-20  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
	(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
	(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
	(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
	(gst_rtp_h263_depay_get_property),
	(gst_rtp_h263_depay_change_state),
	(gst_rtp_h263_depay_plugin_init):
	* gst/rtp/gstrtph263depay.h:
	Added an H263 depayloader. Fixes #369392.

	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
	Make the H263+ pay/depayloader support H263-1998 and H263-2000
	payloads.
	Also alow plain H263 on the h263p payloaders. Fixes #465040.

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2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	Add small comparision with the windowed sinc filters in the docs.

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2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
	(audiochebyshevfreqband_suite):
	* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
	(audiochebyshevfreqlimit_suite):
	Also test 32 bit float mode and the type 2 variants of the filters.

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2007-08-18  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop):
	Refactor the udp and interleaved loop function a bit.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
	(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect connection activity with a new lock, avoids deadlocks when going
	to PAUSED. Fixes #455808.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
	Fix debug statement.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
	Fix stray %u in debug line as spotted by Saur on IRC.

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2007-08-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_class_init):
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_class_init):
	Use generator macros for the process functions for the different
	sample types, add lower upper boundaries for the GObject properties
	so automatically generated UIs can use sliders and add a note about
	the number of poles as a too high number of poles combined with
	very low or very high frequencies will produce only noise.
	* docs/plugins/gst-plugins-good-plugins.args:
	Regenerated for the property changes.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
	(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Improve timeout handling.
	Use the same socket for sending and receiving RTCP packets so that some
	servers can track clients better.
	Improve connection closed handling. Try to reconnect.
	Don't overwrite our content base with NULL.
	Improve debugging.
	Improve range parsing and handling.
	Remove flushing hack now that core does the right thing.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_init), (gst_multiudpsink_set_property),
	(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
	(gst_multiudpsink_close), (gst_multiudpsink_add):
	* gst/udp/gstmultiudpsink.h:
	Add support for getting and setting the socket to use.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_get_property):
	Add support for getting the currently used socket.

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2007-08-16  Sebastian Dröge  <slomo@circular-chaos.org>

	reviewed by: Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_mode_get_type),
	(gst_audio_chebyshev_freq_band_base_init),
	(gst_audio_chebyshev_freq_band_dispose),
	(gst_audio_chebyshev_freq_band_class_init),
	(gst_audio_chebyshev_freq_band_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_band_set_property),
	(gst_audio_chebyshev_freq_band_get_property),
	(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
	(gst_audio_chebyshev_freq_band_start):
	* gst/audiofx/audiochebyshevfreqband.h:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_mode_get_type),
	(gst_audio_chebyshev_freq_limit_base_init),
	(gst_audio_chebyshev_freq_limit_dispose),
	(gst_audio_chebyshev_freq_limit_class_init),
	(gst_audio_chebyshev_freq_limit_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_limit_set_property),
	(gst_audio_chebyshev_freq_limit_get_property),
	(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
	(gst_audio_chebyshev_freq_limit_start):
	* gst/audiofx/audiochebyshevfreqlimit.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
	Fixes #464800.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/audiochebyshevfreqband.c:
	(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
	(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
	* tests/check/elements/audiochebyshevfreqlimit.c:
	(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
	(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
	Add unit tests for the chebyshev filters.

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	And add docs for the chebyshev filters. While doing
	that also run make update in docs/plugins.

Stefan Kost's avatar
Stefan Kost committed
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2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/gstcmmltag.c:
	* gst/rtp/gstrtpvorbispay.c:
	  Make ro memory to share.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Improve UDP performance by avoiding a select() when we have data
	available immediatly.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
	(gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Add (dummy) SSRC management signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
	(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
	(on_timeout), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add connection-speed property.
	Add find_stream helper functions.
	Handle stream EOS based on BYE messages or SSRC timeout.
	Returns SUCCESS from the state change function as we hide our async
	elements from the parent.

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2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/rndbuffersize.c:
	  Fix da leak.

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2007-08-14  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/Makefile.am:
	* gst/debug/breakmydata.c:
	* gst/debug/gstdebug.c:
	* gst/debug/negotiation.c:
	* gst/debug/progressreport.c:
	* gst/debug/rndbuffersize.c:
	* gst/debug/testplugin.c:
	  Add new test element and clean-up the others a little.

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2007-08-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of mp4a version 0 atoms. Fixes #465774.

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2007-08-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpilbcdepay.c:
	  Include stdlib.

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2007-08-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpmpvdepay.c:
	Set the mpegversion in the caps so that autoplugging does not get
	confused.

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2007-08-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/uk.po:
	* po/vi.po:
	  Updated translations.

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2007-08-08  Michael Smith <msmith@fluendo.com>

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Render right border in the correct location.

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2007-08-08  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Olivier Crete <tester at tester dot ca>

	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Make mode property a string. Fixes #464475.

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

	* ext/flac/gstflacenc.c:
	  Widen caps to match decoder a bit and add more FIXMEs.

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

	patch by: Mark Nauwelaerts <manauw@skynet.be>

	* gst/avi/gstavimux.c:
	  Fix ODML index tag numbering. Fixes #463624.

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2007-08-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_udp_sink):
	Fix default clock-rate for realmedia.
	Fix parsing of transport.
	Don't try to link NULL pads.

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2007-07-30  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.skip:
	  Add POTFILES.skip with list of source files that aren't disted at the
	  moment but contain translatable strings. Should hopefully pacify
	  broken tools and make it clearer that these files are left out
	  intentionally (#461600).

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2007-07-30  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
	If the buffer was entirely clipped ... don't try sending it :)

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports):
	If we don't hav a session manager, set the caps on outgoing buffers
	ourselves.
	Force PAUSE/PLAY methods for now until the extensions can overwrite.
	Append final bit of the transport string even when it does not contain a
	placeholder.

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
	(gst_rtsp_ext_list_connect):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
	Clean up the interface list.
	Allow connecting to interface signals for the extensions.
	Remove old extension code.
	Free list on cleanup.
	Allow extensions to send additional RTSP messages.

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Handle a NULL gconf key gracefully by rendering the default element.

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.h:
	Fix include path for extension interface.

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2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.h:
	Also remove a now unecessary variable here.

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2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
	(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
	(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
	(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Don't save format information ourselves, this is already saved in
	GstAudioFilter.

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2007-07-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Use rank to filter out extensions.
	Add url to stream_select interface call.

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2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/base64.c:
	* gst/rtsp/base64.h:
	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
	(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
	(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
	(gst_rtsp_ext_list_setup_media),
	(gst_rtsp_ext_list_configure_stream),
	(gst_rtsp_ext_list_get_transports),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
	(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
	(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspconnection.c:
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c:
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtspmessage.c:
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsprange.c:
	* gst/rtsp/rtsprange.h:
	* gst/rtsp/rtsptransport.c:
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/rtspurl.c:
	* gst/rtsp/rtspurl.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.c:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/test.c:
	Use shiny new RTSP and SDP library.
	Implement RTSP extensions using the new interface.
	Remove a lot of old code.

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2007-07-24  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.

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2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	Don't unref the outgoing buffer twice when dropping it because it's
	outside of the segment.

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2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
	Use the new buffer clipping function from gstaudio here and
	require gst-plugins-base CVS.
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	For framed Wavpack buffers we require a valid timestamp.

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2007-07-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
	(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
	Clip raw audio and video when we can, keep track of current output
	segment.
	Don't leak buffers and events when there is no output pad.
	Improve debugging here and there.

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2007-07-23  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Sync liboil check with plugins-base.

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2007-07-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/Makefile.am:
	  Fix CFLAGS/LIBS.

	* ext/cdio/gstcdiocddasrc.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  Include stdlib

	* ext/cairo/Makefile.am:
	* gst/videofilter/Makefile.am:
	* tests/examples/level/Makefile.am:
	  Use $(LIBM) instead of -lm

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	  Add another example pipeline.

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Alexander Eichner <alexeichi@yahoo.de>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Use define here.

	* sys/v4l2/gstv4l2tuner.c:
	(gst_v4l2_tuner_set_frequency_and_notify):
	  Don't touch the property - its still disabled.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
	(gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
	* sys/v4l2/v4l2src_calls.h:
	  Improve fallback format negotionation. Fixes #451388

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/videocrop.c: (GST_START_TEST):
	  Fix the test.

Stefan Kost's avatar
Stefan Kost committed
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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task),
	(gst_pngdec_sink_setcaps):
	  More docs. More logs in pngdec.

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2007-07-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Initialize num_buffers with minimum value.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame):
	  Handle frame-size query failure gracefully.

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2007-07-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of esds atoms inside mp4a atoms so that we can set correct
	codec_info for AAC audio. Fixes #457097 along with a whole other bunch
	of qt/aac files.

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2007-07-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c:
	(gst_wavpack_dec_clip_outgoing_buffer):
	Fix buffer clipping to correctly clip to the segment stop.

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2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* tests/Makefile.am:
	Remove bogus check for libcheck, since we check for
	gstreamer-check and it pulls in the required info from there,
	and we weren't actually _using_ the information for libcheck
	ourselves anyway.

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2007-07-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Use pkg-config to locate check.

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2007-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	* gst/debug/gstnavigationtest.c: (gst_navigationtest_transform):
	* gst/effectv/gstaging.c: (gst_agingtv_transform):
	* gst/effectv/gstdice.c: (gst_dicetv_transform):
	* gst/effectv/gstedge.c: (gst_edgetv_transform):
	* gst/effectv/gstquark.c: (gst_quarktv_transform):
	* gst/effectv/gstrev.c: (gst_revtv_transform):
	* gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform):
	* gst/effectv/gstvertigo.c: (gst_vertigotv_transform):
	* gst/effectv/gstwarp.c: (gst_warptv_transform):
	* gst/matroska/matroska-demux.c:
	(gst_matroska_demux_add_wvpk_header),
	(gst_matroska_demux_check_subtitle_buffer),
	(gst_matroska_decode_buffer):
	* gst/videofilter/gstvideoflip.c: (gst_video_flip_transform):
	  Fix build against core CVS.

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2007-07-10  Edward Hervey  <bilboed@gmail.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
	don't have enough granularity to convert that boolean into a
	GstFlowReturn.

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2007-07-06  Michael Smith <msmith@fluendo.com>

	* gst/law/alaw-decode.c: (alawdec_sink_setcaps),
	(gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain),
	(gst_alawdec_change_state):
	* gst/law/alaw-decode.h:
	* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	(gst_mulawdec_class_init), (gst_mulawdec_init),
	(gst_mulawdec_chain), (gst_mulawdec_change_state):
	* gst/law/mulaw-decode.h:
	  Fix capsnego bogosity in *law decoders. 

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2007-07-06  Michael Smith <msmith@fluendo.com>

	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
	(gst_smokeenc_setcaps), (gst_smokeenc_chain),
	(gst_smokeenc_change_state):
	* ext/jpeg/gstsmokeenc.h:
	  Remove stupidity in get/set caps functions.
	  Fix some refcounting problems.

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2007-07-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
	Remove endianness-flipping hack that seems to have been required
	only because of a bug in ffmpegcolorspace.
	Partially Fixes: #451908

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2007-07-05  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

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2007-07-03  Wim Taymans,,,  <set EMAIL_ADDRESS environment variable>

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

	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Set the encoding-name in the rtp caps to all uppercase, as required by
	the caps spec.
	Some small cleanups in the error paths. Fixes #453037.

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

	* ext/wavpack/gstwavpackparse.c:
	(gst_wavpack_parse_index_get_last_entry),
	(gst_wavpack_parse_index_get_entry_from_sample),
	(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	(gst_wavpack_parse_scan_to_find_sample):
	* ext/wavpack/gstwavpackparse.h:
	Use a GSList for the GArray that is used like a list anyway.

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2007-06-28  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
	(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
	  Add state change function where we set 0/1 as default framerate in
	  case our setcaps function isn't called, like it might not in a
	  filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
	  gdkpixbufdec trying to create caps with a 0/0 framerate.
	  Also post an error message on the bus if gst_pad_push() fails when
	  called from our sink event handler (+1 for flow returns for event
	  functions in 0.11) instead of failing silently.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps):
	Cast stack args to the proper types. Fixes #451249.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (gst_rtspsrc_setup_streams):
	* gst/rtsp/gstrtspsrc.h:
	For container formats we only need to activate one of the streams so
	that we correctly signal no-more-pads. Fixes #451015.

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

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update docs with caps info.

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

	* po/POTFILES.in:
	  Add more files with translatable strings (#450878).

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

	* MAINTAINERS:
	Updating all the maintainers files

Edward Hervey's avatar
Edward Hervey committed
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2007-06-22  Edward Hervey  <edward@fluendo.com>

	* ext/flac/gstflactag.c: (gst_flac_tag_init):
	* gst/interleave/deinterleave.c: (deinterleave_init),
	(deinterleave_sink_link):
	* gst/interleave/interleave.c: (interleave_init):
	* gst/median/gstmedian.c: (gst_median_init):
	* gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init):
	Fix memory leaks.
	* tests/check/elements/id3demux.c: (pad_added_cb):
	Remove unused variable.

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2007-06-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.h:
	Make the prototype of gst_gconf_get_key_for_sink_profile
	match the implementation.
	Patch by: Damien Carbery <damien dot carbery at sun dot com>
	Fixes: #449747

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2007-06-20  Michael Smith <msmith@fluendo.com>

	* gst/rtp/gstrtpdepay.c:
	  Fix description - rtpdepay is not a payloader.

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2007-06-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
	(qtdemux_video_caps):
	* gst/qtdemux/qtdemux_fourcc.h:
	  Add MJPG to the variants of motion jpeg.

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2007-06-19  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
	* tests/check/elements/videocrop.c: (GST_START_TEST):
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	* tests/check/elements/wavpackparse.c: (GST_START_TEST):
	  Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
	  error flags are included and it errors out on compiler warnings
	  for CVS builds; remove unused variables in various unit tests.

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2007-06-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close), (rtsp_connection_free):
	Use threadsafe inet_ntop to convert an ip number to a string. 
	Fixes #447961.
	Don't leak fd (and ip) when freeing a connection without first closing
	it.

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2007-06-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.6 to the doap file.

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=== release 0.10.6 ===

2007-06-18  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.6, "Wobble Board"

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2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  Revert previous commit again, since we are frozen (sorry).

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2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  inet_ntoa() uses a static buffer internally, so we need to copy the
	  returned string if we want to store it for later (#447961).

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2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* win32/vs6/autogen.dsp:
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstalaw.dsp:
	* win32/vs6/libgstalpha.dsp:
	* win32/vs6/libgstalphacolor.dsp:
	* win32/vs6/libgstapetag.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstauparse.dsp:
	* win32/vs6/libgstautodetect.dsp:
	* win32/vs6/libgstavi.dsp:
	* win32/vs6/libgstcutter.dsp:
	* win32/vs6/libgstdirectdraw.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgsteffectv.dsp:
	* win32/vs6/libgstflx.dsp:
	* win32/vs6/libgstgoom.dsp:
	* win32/vs6/libgsticydemux.dsp:
	* win32/vs6/libgstid3demux.dsp:
	* win32/vs6/libgstinterleave.dsp:
	* win32/vs6/libgstjpeg.dsp:
	* win32/vs6/libgstlevel.dsp:
	* win32/vs6/libgstmatroska.dsp:
	* win32/vs6/libgstmedian.dsp:
	* win32/vs6/libgstmonoscope.dsp:
	* win32/vs6/libgstmulaw.dsp:
	* win32/vs6/libgstmultipart.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsmpte.dsp:
	* win32/vs6/libgstspeex.dsp:
	* win32/vs6/libgstudp.dsp:
	* win32/vs6/libgstvideobalance.dsp:
	* win32/vs6/libgstvideobox.dsp:
	* win32/vs6/libgstvideocrop.dsp:
	* win32/vs6/libgstvideoflip.dsp:
	* win32/vs6/libgstvideomixer.dsp:
	* win32/vs6/libgstwaveform.dsp:
	* win32/vs6/libgstwavenc.dsp:
	* win32/vs6/libgstwavparse.dsp:
	Mark *.dsp & *.dsw as binary files and convert to DOS line
	endings, as they don't load into VS6 correctly otherwise.

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2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect):
	Fix the MingW build. 
	Patch By: Vincent Torri <vtorri at univ-evry dot fr>
	Fixes: #446981

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2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/.cvsignore:
	* tests/icles/.cvsignore:
	Hush the buildbots up

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2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/directdraw/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/waveform/Makefile.am:
	Make sure to dist everything needed for win32 builds.

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2007-06-14  Edward Hervey  <edward@fluendo.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	For AMR-NB streams, export the AMRSpecificBox as codec_data on the
	caps.
	Fixes #447458

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

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	Make sure we allocate enough memory for the codec_data.
	Fixes #447210.

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2007-06-12  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add videocrop project file to the win32 manifest.
	* win32/vs6/gst_plugins_good.dsw:
	Add qtdemux,videocrop and waveform projects to the workspace.
	* win32/vs6/libgstqtdemux.dsp:
	Add zlib to the link list of qtdemux.
	* win32/vs6/libgstvideocrop.dsp:
	Add a project file for videocrop.

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* po/POTFILES.in:
	Add qtdemux for translation

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* gst-plugins-good.spec.in:
	* sys/Makefile.am:
	* tests/check/Makefile.am:
	* tests/icles/Makefile.am:
	* tests/icles/videocrop-test.c:
	Move videocrop and osxvideo from -bad.

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-qtdemux.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* win32/MANIFEST:
	Move qtdemux from -bad.

	* gst-plugins-good.spec.in:
	Update spec file to reflect moving of qtdemux and wavpack

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>
	
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	* win32/MANIFEST:
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	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-directdraw.xml:
	* docs/plugins/inspect/plugin-directsound.xml:
	* docs/plugins/inspect/plugin-waveform.xml:
	Move the waveform plugin from -bad too. Update the inspect xml
	files to mention Plugins Good instead of Plugins Bad.

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2007-06-12  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
	(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
	finalization and resuscitation. No longer public.
	(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
	(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
	(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
	(gst_v4l2_buffer_pool_destroy): Make the pool follow common
	miniobject semantics, and be threadsafe.
	(gst_v4l2src_queue_frame): Remove this function, as we just call
	the ioctls directly in the two places where we queue buffers.
	(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
	directly.
	(gst_v4l2src_capture_init): Use the new buffer_pool_new function
	to allocate the pool, which also preallocates the GstBuffers.
	(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
	queueing the frames directly.
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	(gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
	mmap buffers have been dequeued.
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	* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
	real MiniObject instead of rolling our own refcounting and
	finalizing. Give it a lock.
	(struct _GstV4l2Buffer): Remove one intermediary object, having
	the buffers hold the struct v4l2_buffer directly.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
	capture_init so that it can set them on the buffers that it will
	create.
	(gst_v4l2src_get_read): For better or for worse, include the
	timestamping and offsetting code here; really we should be using
	bufferalloc though.
	(gst_v4l2src_get_mmap): Just make grab_frame return one of our
	preallocated, mmap'd buffers.

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

	Patch by: daniel fischer <dan at f3c dot com>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
	(gst_ximage_src_get_caps):
	Actually use the display_name property so that we can dump any
	available X display. Fixes #445905.

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

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

	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
	Add missing rate fields to caps. Fixes #441118.

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2007-06-10  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs8/gst-plugins-good.sln:
	Add DirectSound and DirectDraw sinks project files to
	workspace and solution files.

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2007-06-10  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Josh Coalson <xflac at yahoo dot com>,
	updated by Alexis Ballier <aballier at gentoo dot org>:

	* configure.ac:
	* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
	(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
	(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
	(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
	(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
	(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
	(gst_flac_enc_chain), (gst_flac_enc_set_property),
	(gst_flac_enc_get_property), (gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
	
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2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
	Remove workaround for bug #421543. This is fixed in core 0.10.13 and
	not necessary anymore as we need at least that core version. 

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2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain):
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	(gst_wavpack_parse_push_buffer):
	* ext/wavpack/gstwavpackparse.h:
	Improve discont handling by checking if the next Wavpack block has
	the expected, following block index.

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2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
	  Fix element description.

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2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* ext/Makefile.am:
	* tests/check/Makefile.am:
	  move wavpack plugin.  See #352605.

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2007-06-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* sys/Makefile.am:
	* win32/MANIFEST:
	Add DirectDraw & DirectSound plugins to the build and docs.

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2007-06-08  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
	* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
	  When operating in pull mode, error out correct on not-linked.

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

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
	(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
	format and size if the ioctls are defined; should fix compilation
	on Linux < 2.16.19.

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

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Printf fixes in debug statements; use LOG level for debug statements
	  that are printed for each and every frame; convert c++ comments to
	  C-style comments; not much point using g_try_malloc() if we then not
	  even check the return value.

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

	* configure.ac:
	  Bump requirements to released versions (core and base 0.10.13).

	* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  own implementation.

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2007-06-05  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
	some useless comments.

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
	frames before calling STREAMON, that might leave them in a state
	where they can't be dequeued if we go back to NULL without calling
	STREAMON, according to the docs.
	(gst_v4l2src_capture_start): Enqueue buffers here instead, right
	before we call STREAMON.
	(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
	failures. (For me this code hung.) The pool refcounting is still
	crack; added a note to that effect.

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

	* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
	(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
	Add support for mapping gst structure names to the MIME type equivalent.
	Implemented for audio/x-mulaw->audio/basic. Fixes #442874.

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2007-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),</