ChangeLog 1.21 MB
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2007-09-05  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
	(gst_sdp_message_init), (gst_sdp_message_uninit),
	(is_multicast_address), (gst_sdp_message_as_text),
	(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
	(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
	(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
	(gst_sdp_media_init), (gst_sdp_media_uninit),
	(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
	(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
	(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
	(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
	(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Separate INIT_ARRAY() and related macros into two versions, one for
	structures and one for pointers (e.g., INIT_ARRAY() and
	INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
	lists of emails and phone numbers.
	Add missing const as appropriate.
	Change all gint to guint since they all actually represent unsigned
	values.
	Do not use time as a variable name as it shadows the global time().
	Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
	Actually implement gst_sdp_message_add_time().
	Make gst_sdp_message_add_time() take repeat times as an argument.
	Store repeat times in GstSDPTime as a GArray rather than as gchar**.
	Corrected the definition of gst_sdp_media_get_bandwidth() (was
	misspelled as badwidth).
	gst-indented and a little clean up. Fixes #471067.

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2007-09-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_process_double), (volume_process_double_clamp),
	(volume_process_float_clamp):
	Correctly clamp float/double samples in the [-1.0,1.0] range to
	prevent weird effects.
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	Add unit tests for all samples types that had none before.

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

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Need to include stdlib.h for abs() here too.

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

	* gst/playback/gststreaminfo.c:
	  Fix build.

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2007-09-05  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gststreaminfo.c:
	  Clean up some half-disabled code and comment.

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

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	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_payload_audio_handle_event):
	Return FALSE from the event handler to let the parent class handle the
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	event. Fixes #446766.
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	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
	Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	Bump the MTU to 1400.

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2007-09-03  Johan Dahlin  <jdahlin@async.com.br>

	* gst/typefind/gsttypefindfunctions.c (plugin_init): 
	Add an audio/x-nsf typefind function for the nsfdec element.

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2007-09-03  Renato Filho  <renato.filho@indt.org.br>
	* gst/playback/gstplaybasebin.c:
	Included "myth://" on stream_uris list for enable buffering to mythtv files

Wim Taymans's avatar
Wim Taymans committed
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2007-09-03  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
	(gst_rtcp_unix_to_ntp):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix parsing of RB blocks.
	Fix docs.
	Added helper functions to convert to/from UNIX and NTP time.
	API: gst_rtcp_ntp_to_unix()
	API: gst_rtcp_unix_to_ntp()

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
	(gst_rtp_buffer_get_header_len),
	(gst_rtp_buffer_get_extension_data),
	(gst_rtp_buffer_get_payload_subbuffer),
	(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
	(gst_rtp_buffer_ext_timestamp):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Fix some more docs.
	Implement handling of packets with extensions.
	Fix padding check in _validate().
	Added function to get extension data.
	API: gst_rtp_buffer_get_header_len()
	API: gst_rtp_buffer_get_extension_data()

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

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_class_init),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Add some more docs for the queue-delay property and fix a typo in a
	comment.

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	Fix typo.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_change_state):
	When skew slaving, try to hover around the middle of a segment so that
	we at most drift by half a segment.
	If we are aligning in the oposite direction of the clock skew, we don't
	have to resync.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Be less silly with the segment start, just apply the clock-base to the
	timestamp.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_class_init),
	(gst_base_rtp_depayload_finalize),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Deprecate the queue handling thread thing and remove the code.
	Use new method to calculate the extended timestamp.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_packet_sdes_copy_entry):
	Use g_strndup which does exactly what we want.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
	(gst_rtp_buffer_ext_timestamp):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Add helper function to compare seqnums.
	Add helper function to calculate extended timestamps.
	API: gst_rtp_buffer_compare_seqnum()
	API: gst_rtp_buffer_ext_timestamp()

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2007-08-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_packet_sdes_get_entry),
	(gst_rtcp_packet_sdes_copy_entry):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix and document SDES item data function.
	Add new function that makes a proper copy of SDES item data.
	API: gst_rtcp_packet_sdes_copy_entry()

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/Makefile.am:
	  The tcp and subparse plugins are under gst, but not totaly free of
	  dependencies. Handle selection inconfigure.ac, so that they show up
	  on the final list of what is build and what is not. Maybe they should
	  better be moved to ext.

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Daniel Díaz  <yosoy@danieldiaz.org>

	* configure.ac:
	* gst/Makefile.am:
	  Check if libxml provides HTML parser which subparse needs.
	  Fixes #451970.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c:
	  Fix typo and compilation on big endian systems.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstssaparse.c:
	  Convert SSA newline codes into actual newline characters (#470766).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* tests/check/libs/pbutils.c:
	  API: also add gst_install_plugins_supported() while we're at it
	  (see #470456).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/missing-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.h:
	* tests/check/libs/pbutils.c:
	  API: add gst_missing_*_installer_detail_new() convenience API so
	  that applications that know exactly what they're missing can request
	  installer detail strings for those items directly instead of having
	  to first create a dummy missing-plugin message and then get the
	  installer detail string from that.  Fixes #470456.

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

	* gst/playback/gstdecodebin.c: (close_pad_link):
	We need to set up delayed-linking whenever the caps are non-fixed,
	not just when there are multiple types - use gst_pad_is_fixed()
	to test.

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2007-08-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_plugin_message_get_installer_detail):
	  Add missing separator in PID fallback case.

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2007-08-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/alsa/Makefile.am:
	There is no GST_PLUGINS_BASE_LIBS defined.
	
	* ext/alsa/gstalsa.c:
	* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
	Add support for ALSA 24-bit formats.
	snd_pcm_delay can return an error code, especially
	during XRUNS. In that case, the best we can do is assume
	delay = 0.

	* gst/audioconvert/Makefile.am:
	Add flags from -base before any more-remote dependencies.

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2007-08-23  Sebastian Dröge  <slomo@circular-chaos.org>

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	Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
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	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_set_volume),
	(gst_volume_init), (volume_process_int32),
	(volume_process_int32_clamp), (volume_process_int24),
	(volume_process_int24_clamp), (volume_process_int16),
	(volume_process_int16_clamp), (volume_process_int8),
	(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
	* gst/volume/gstvolume.h:
	Add support for int32, int24 and int8 to the volume element.
	Fixes #445529.

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2007-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/examples/Makefile.am:
	  Fix even more.

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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* ext/gnomevfs/gstgnomevfssrc.c:
	* ext/gnomevfs/gstgnomevfssrc.h:
	* gst-libs/gst/Makefile.am:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	* sys/v4l/v4lsrc_calls.c:
	* tests/examples/Makefile.am:
	* win32/common/config.h:
	  Revert unwanted commit. many thanks to moap. I want a fix for 
	  https://thomas.apestaart.org/moap/trac/ticket/239

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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  Move passthrough below gst_object_sync_values(). Fixes #442654.

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

	* gst-libs/gst/audio/audio.c:
	Clarify the docs a little.

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

	* gst/volume/gstvolume.c:
	  Enable liboil for float and add more details about problems with
	  int16.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
	Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	When calculating the first timestamp of the buffers, don't go below 0
	and clip the samples because the offset was on the eos page.
	Fixes #466717.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
	(gst_ogg_demux_collect_chain_info):
	Also submit the eos page when trying to find the first timestamp.
	See #466717.

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

	* gst-libs/gst/audio/audio.h:
	Use gst_util_uint64_scale() instead of doing the math
	with double for GST_FRAMES_TO_CLOCK_TIME() and
	GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
	prevents rounding errors. Fixes #467667.

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

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
	(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	Small cleanups.
	On shutdown, don't read the control socket yet.
	Set timeout value correctly in all cases.
	Add function to check if the server accepts reads or writes.
	API: gst_rtsp_connection_poll()

	* gst-libs/gst/rtsp/gstrtspdefs.h:
	Fix compilation with -pedantic.
	Add enum for _poll.

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

	Patch by: Olivier Crete  <tester at tester ca>

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
	(gst_basertppayload_getcaps):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Add getcaps vfunc to basertppayload. See #465146.

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

	* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
	Only post buffering messages when we are a stream.

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2007-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.c:
	  Small docs fix and addition.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-textoverlay.c:
	  Add a dumb little test for textoverlay alignments.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams  <dcbw redhat com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  API: add "line-alignment" property (#459334). Add gtk-doc blurb for
	  "silent" property so there's a Since tag in the API reference.

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

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_set_outcaps):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Improve caps negotiation so that downstream elements can confiure
	certain RTP properties by fixing them on the caps. See #465146.
	Add docs.

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2007-08-11  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Mark as deprecated some macros which were presumably meant to be
	  private API and accidentally exposed in the public header file.
	  Also actually _init() lock (only works at the moment because the
	  struct is zeroed out when created and the initial values in the
	  mutex struct are zeroes too). (#459585)

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

	* docs/libs/Makefile.am:
	  Remove cruft and do some cleanups.

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	  Prepare for comming gtkdoc features (rebase against online docs).

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	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Debug output fixes.
	* tests/check/elements/audiorate.c: (do_perfect_stream_test),
	(GST_START_TEST):
	  Change the number of buffers used; 500 is too many and leads to
	  timeouts.

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2007-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstqueue2.c:
	* gst/videorate/gstvideorate.c:
	  Printf format fixes (#465028).

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2007-08-09  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If we have a large (> 1 second) discontinuity, push a series of
	  smaller buffers rather than a single very large buffer. Avoids
	  unreasonably large single buffer allocations when encountering a
	  large gap.
	* tests/check/elements/audiorate.c: (GST_START_TEST),
	(audiorate_suite):
	  Add a test for this.

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2007-08-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybasebin.c: (group_commit),
	(queue_remove_probe), (queue_threshold_reached):

	Patch by: Josep Torra Valles <josep@fluendo.com>
	Fixes: #465015
	Make sure we remove the check_queues buffer probe from the 
	correct queue to avoid racily going back to "buffering 99%" when
	buffering is actually complete.

	Also, fix the spelling of Josep's surname in the ChangeLog.

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2007-08-09  Stefan Kost  <ensonic@users.sf.net>

	* ext/ogg/gstoggmux.c:
	  Do not leak oggmux instance.
	
	* ext/vorbis/vorbisenc.c:
	  Also log values.

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

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

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	patch by: Yang Hong <hongyang@redflag-linux.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  Add 'silent' property to GstTimeOverlay. Fixes #462979

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	Patch by: Josep Torra Valles <josep@fluendo.com>
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	* docs/plugins/gst-plugins-base-plugins.args:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
	(gst_uri_decode_bin_get_property), (gen_source_element):
	Add connection-speed property. Fixes #464690.

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2007-08-07  Wim Taymans  <wim.taymans@gmail.com>

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

	* configure.ac:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect):
	Fix compilation on windows. Fixes #464320.

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	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (queue_threshold_reached),
	(gen_source_element), (setup_substreams),
	(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
	(gst_play_base_bin_get_streaminfo_value_array):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_set_property), (gst_play_bin_get_property),
	(gst_play_bin_handle_redirect_message):
	Move connection-speed property from playbin to playbasebin so that we
	can also configure it in source elements that have the connection-speed
	property. Fixes #464028.
	Add some debug info here and there.

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2007-08-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
	Properly respond to conversion queries. Fixes #464079.

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2007-08-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
	(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
	(gst_audio_test_src_init_sine_table),
	(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
	* gst/audiotestsrc/gstaudiotestsrc.h:
	Add float/double and int32 support to audiotestsrc. Fixes #460422.
	Also set the default volume to the default value specified in the
	GParamSpec.

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2007-08-03  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Jens Granseuer <jensgr at gmx dot net>

	* gst/audioconvert/gstaudioquantize.c:
	Fix C89 incompatibilities and spelling of explanations. Fixes #463215.

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

	* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
	Add rdt manager for rdt transport.
	Fix parsing of RDT transport.

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

	* configure.ac:
	Back to CVS

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=== release 0.10.14 ===

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

	* configure.ac:
	  releasing 0.10.14, "Light Years Ahead"

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	* tests/check/libs/audio.c: (GST_START_TEST):
	Fix the test to reflect the behaviour of gst_audio_clip_buffer.

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

	* gst-libs/gst/audio/audio.c:
	When clipping a buffer with no timestamp, assume it is
	within the segment without warnings.

	Fixes: #460978

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

	* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
	Fire the signal on the object, not the interface.

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

	* gst-libs/gst/rtsp/.cvsignore:
	Ber. Don't include the full path, idiot.

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

	* gst-libs/gst/rtsp/.cvsignore:
	Ignore generated files.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/interfaces-marshal.list:
	* gst-libs/gst/interfaces/rtspextension.c:
	* gst-libs/gst/interfaces/rtspextension.h:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtsp.h:
	* gst-libs/gst/rtsp/gstrtspextension.c:
	(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_detect_server),
	(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	(gst_rtsp_extension_configure_stream),
	(gst_rtsp_extension_get_transports),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/rtsp/gstrtspextension.h:
	* gst-libs/gst/rtsp/rtsp-marshal.list:
	Move the rtspextension.h interface into gstrtspextension.h
	as part of libgstrtsp instead of libgstinterfaces, because it's
	only for use within plugins, not applications. 
	Add stuff to do the enum & marshal generation needed in libgstrtsp now.
	Use the GST_TYPE_RTSP_RESULT enum type for the return value of the 
	signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
	is abstract.

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

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/interfaces-marshal.list:
	* gst-libs/gst/interfaces/rtspextension.c:
	(gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/interfaces/rtspextension.h:
	Fix marshaller for the send signal.
	Add URL to stream selection interface method.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/riff/Makefile.am:
	Pull in our dependencies from -base before those from outside.

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	API: gst_rtsp_base64_decode_ip()
	Added function to decode Base64 in-place.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/libs/.cvsignore:
	Ignore the mixer test binary.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	Gratuitous comment change to trigger a rebuild on the buildbots.

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

	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
	(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
	(gst_sdp_media_get_format), (gst_sdp_media_get_information),
	(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
	(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
	(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Constify args where we can.

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

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/rtspextension.c:
	(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_detect_server),
	(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	(gst_rtsp_extension_configure_stream),
	(gst_rtsp_extension_get_transports),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/interfaces/rtspextension.h:
	Move interface for RTSP extensions from -good to here.
	Added helper methods to invoke interface methods.

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
	(gst_rtsp_message_init_response),
	(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
	(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
	(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	(gst_rtsp_message_get_body), (dump_key_value):
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse):
	* gst-libs/gst/rtsp/gstrtsprange.h:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst-libs/gst/rtsp/gstrtspurl.c:
	Fix some more RTSP docs.
	Add some missing methods for dealing with messages.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect), (add_auth_header),
	(gst_rtsp_connection_write), (gst_rtsp_connection_send),
	(read_body), (gst_rtsp_connection_receive),
	(gst_rtsp_connection_next_timeout),
	(gst_rtsp_connection_reset_timeout),
	(gst_rtsp_connection_set_auth):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse):
	* gst-libs/gst/rtsp/gstrtspurl.h:
	Added beginnings of RTSP documentation.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/sdp/gstsdp.h:
	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
	(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
	(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val),
	(gst_sdp_message_add_attribute), (gst_sdp_media_new),
	(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
	(gst_sdp_media_get_media), (gst_sdp_media_set_media),
	(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
	(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
	(gst_sdp_media_get_format), (gst_sdp_media_add_format),
	(gst_sdp_media_get_information), (gst_sdp_media_set_information),
	(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
	(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
	(gst_sdp_media_set_key), (gst_sdp_media_get_key),
	(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
	(gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
	(print_media), (gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Document the SDP library.
	Add some of the missing SDPMedia methods.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
	(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
	(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
	(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
	(parse_response_status), (parse_request_line), (parse_line),
	(gst_rtsp_connection_read), (read_body),
	(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
	(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
	(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
	(gst_rtsp_connection_set_auth):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
	(gst_rtsp_strresult), (gst_rtsp_method_as_text),
	(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
	(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
	(gst_rtsp_find_method):
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	(gst_rtsp_message_new), (gst_rtsp_message_init),
	(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
	(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
	(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
	(gst_rtsp_message_free), (gst_rtsp_message_add_header),
	(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
	(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
	(gst_rtsp_message_dump):
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse), (gst_rtsp_range_free):
	* gst-libs/gst/rtsp/gstrtsprange.h:
	* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
	(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
	(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
	(range_as_text), (rtsp_transport_mode_as_text),
	(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
	(gst_rtsp_transport_free):
	* gst-libs/gst/rtsp/gstrtsptransport.h:
	* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
	(gst_rtsp_url_free), (gst_rtsp_url_set_port),
	(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
	* gst-libs/gst/rtsp/gstrtspurl.h:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/sdp/gstsdp.h:
	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
	(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
	(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
	(gst_sdp_attribute_init), (gst_sdp_message_new),
	(gst_sdp_message_init), (gst_sdp_message_uninit),
	(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
	(gst_sdp_media_uninit), (gst_sdp_media_free),
	(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
	(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
	(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
	(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
	(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val),
	(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
	(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
	(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
	(gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
	(read_string), (read_string_del), (gst_sdp_parse_line),
	(gst_sdp_message_parse_buffer), (print_media),
	(gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	Move SDP and RTSP from helper objects in -good to a reusable library.
	Use a proper gst_ namespace.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
	(vorbis_dec_flush_decode):
	Use the new buffer clipping function from gstaudio here.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
	* gst-libs/gst/audio/audio.h:
	* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
	API: Add buffer clipping function for raw audio buffers. Fixes #456656.
	Also add deprecation guards for gst_audio_structure_set_int() to the
	header.

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Cleanup the docs.

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

	Patch by: Dan Williams <dcbw at redhat dot com>

	* gst/playback/gstplaybasebin.c:
	(gst_play_base_bin_get_streaminfo_value_array):
	Don't return NULL when querying the stream info value array but instead
	return an empty array. Fixes #459204.

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2007-07-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gsturidecodebin.c:
	  Init debug category before using it.

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2007-07-21  Jan Schmidt  <thaytan@noraisin.net>

	* gst-libs/gst/interfaces/mixer.h:
	Add padding vars in place of the signal pointers
	when building with DISABLE_DEPRECATED so that the
	interface structure doesn't change size.

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2007-07-20  Jan Schmidt  <thaytan@noraisin.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixerelement.c:
	* ext/alsa/gstalsamixertrack.c:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/interfaces/mixer.h:
	* gst-libs/gst/interfaces/mixeroptions.c:
	* gst-libs/gst/interfaces/mixeroptions.h:
	* gst-libs/gst/interfaces/mixertrack.c:
	* gst-libs/gst/interfaces/mixertrack.h:
	* tests/check/Makefile.am:
	* tests/check/libs/mixer.c:

	Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
	Fixes: #152864 

	Add support for notifying mixer changes on the message bus, and
	implement it in alsamixer.

	API: gst_mixer_get_mixer_flags
	API: gst_mixer_message_parse_mute_toggled
	API: gst_mixer_message_parse_record_toggled
	API: gst_mixer_message_parse_volume_changed
	API: gst_mixer_message_parse_option_changed
	API: GstMixerMessageType
	API: GstMixerFlags

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2007-07-20  Michael Smith <msmith@fluendo.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
	  xcontext->im_format is only for testing XShm support (as the header
	  file comments document). Use xvimage->im_format for everything else.
	  Avoids spurious warnings on buffer allocation before setcaps.

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

	* tests/examples/volume/Makefile.am:
	* tests/icles/Makefile.am:
	  We should use $(LIBM).

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

	* tests/icles/Makefile.am:
	  This needs -lm.

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

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_class_init), (gst_basertppayload_init),
	(gst_basertppayload_set_property),
	(gst_basertppayload_get_property):
	Don't break ABI, restore previous ranges. Keep the default random
	selection of timestamp and seqnum offset but as soon as the app sets a
	specific value, use that one.

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Bastien Nocera <hadess at hadess dot net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init):
	* sys/xvimage/xvimagesink.h:
	Add option to turn off double-buffering for debugging purposes.
	Fixes #437169.

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Jorn Baayen <jorn at openedhand dot com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
	(gst_ximagesink_set_property), (gst_ximagesink_get_property),
	(gst_ximagesink_init), (gst_ximagesink_class_init):
	* sys/ximage/ximagesink.h:
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init):
	* sys/xvimage/xvimagesink.h:
	add 'handle-expose' property. Useful for video widgets which may want to
	be in control of Expose behaviour. Fixes #380625

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2007-07-14  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_class_init), (gst_basertppayload_init),
	(gst_basertppayload_event), (gst_basertppayload_push),
	(gst_basertppayload_set_property),
	(gst_basertppayload_get_property),
	(gst_basertppayload_change_state):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Fix ranges of rtp payloader properties so that the full range can be
	used in addition to -1 (random).
	Fix wrong seqnum reporting in caps.
	Fixes #420326.

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2007-07-13  Wim Taymans  <wim.taymans@gmail.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_init),
	(gst_video_rate_query):
	Use boilerplate.
	Add latency query, might not be perfect yet but already works a lot
	better. Fixes #442557.

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

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_setcaps):
	* sys/xvimage/xvimagesink.h:
	After a caps change, redraw our borders to avoid garbage left there
	when the image format changes to a smaller size, like 16:9 -> 4:3
	Also, hold the flow_lock a bit longer in the set_caps while we're
	fiddling with the xcontext.

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

	* Makefile.am:
	* 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.