ChangeLog 451 KB
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2007-06-01  Wim Taymans  <wim@fluendo.com>

	Based on Patch by: Daniel Charles <dcharles at ti dot com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
	(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
	(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpamrpay.h:
	Add support for AMR-WB.
	Small cleanups such as using BOILERPLATE.

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2007-05-31  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
	Fix compile warning when debug is disabled as spotted bu Saur on IRC.

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2007-05-30  Andy Wingo  <wingo@pobox.com>

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	* sys/v4l2/gstv4l2object.h: 
	* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
	unintended changes.

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	* sys/v4l2/v4l2src_calls.h: 
	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
	the format list in the order that the driver gives it to us.
	(gst_v4l2src_probe_caps_for_format_and_size)
	(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
	based on the capabilities of the device.
	(gst_v4l2src_grab_frame): Update for object variable renaming.
	(gst_v4l2src_set_capture): Update to be strict in its parameters,
	as in the set_caps below.
	(gst_v4l2src_capture_init): Update for object variable renaming,
	and reflow.
	(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
	(gst_v4l2src_capture_deinit): Update for object variable renaming.
	(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
	(gst_v4l2src_get_fps): Remove; these functions don't have much
	meaning outside of an atomic set_caps method.
	(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
	known.

	* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
	call to update_fps; not sure about this change.
	(gst_v4l2_tuner_set_norm): Work around the fact that for the
	moment we don't have an update_fps_func.

	* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
	structures in the object, just store what we need. Do store the
	probed caps of the device. Don't store the current frame rate.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
	update_fps_function, for now. Update for new object variable
	naming.
	(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
	new object variable naming.
	(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
	(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
	(gst_v4l2src_get_caps): Rework to probe the device for supported
	frame sizes and frame rates.
	(gst_v4l2src_set_caps): Rework to be strict in the given
	parameters: if someone asks us to have a certain size and rate,
	that is what we configure.
	(gst_v4l2src_get_read): Update for object variable naming. Don't
	leak buffers on short reads.
	(gst_v4l2src_get_mmap): Update for object variable naming, and add
	comments.
	(gst_v4l2src_create): Update for object variable naming.

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2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
	(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
	* gst/avi/gstavidemux.h:
	  Parse subtitle text streams instead of erroring out (#442034). Still
	  needs a parser for the subtitles to actually show up.

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2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
	(gst_avi_demux_loop):
	  Make _push_event() return TRUE if the event could be pushed on at
	  least one pad and not only if it could be pushed on all pads,
	  otherwise we'll end up posting an error message on EOS if one or
	  more source pads are not connected.

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2007-05-28  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Use renamed RTP bin.

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2007-05-28  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>

	* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	(gst_video_box_set_property), (gst_video_box_transform_caps),
	(video_box_recalc_transform), (gst_video_box_set_caps),
	(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
	(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
	(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
	(gst_video_box_i420_i420), (gst_video_box_transform),
	(plugin_init):
	Add AYUV->AYUV and AYUV->I420 formats. 
	Fix negotiation and I420->AYUV conversion.
	Fixes #429329.

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2007-05-26  Wim Taymans  <wim@fluendo.com>

	* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	Use different variables for nested for loops so that the outer loop
	functions properly and speex files with multiple frames per buffer work
	properly.
	Fixes #441408.

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2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
	  Don't leak newsegment events.

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2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
	  drags it in.

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2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(notgst_value_array_append_buffer),
	(gst_flac_enc_process_stream_headers),
	(gst_flac_enc_write_callback), (gst_flac_enc_chain),
	(gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	  Collect headers, add "streamheader" field to output caps and set
	  BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
	  produces output according to the official FLAC-to-Ogg mapping
	  instead of completely broken files. Fixes #426044.

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2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
	(gst_id3demux_send_new_segment), (gst_id3demux_chain),
	(gst_id3demux_sink_event):
	* gst/id3demux/gstid3demux.h:
	* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
	(gst_tag_demux_chain), (gst_tag_demux_sink_event),
	(gst_tag_demux_send_new_segment):
	Handle and adjust new-segment events so that downstream really
	sees a stream with the tag pieces stripped off the front and back.
	Fixes strangeness in seeking when mp3 decoders use the new-segment
	byte position to estimate their current playback position timestamp
	and then the arriving buffers don't match up.

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2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
	  Don't unnecessarily perform a READY->NULL->READY transition on the
	  detected audio sink when starting up. Fixes: #440127

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2007-05-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
	(gst_flac_enc_chain):
	  Don't crash in chain function if setcaps hasn't been called.

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2007-05-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
	Init value to avoid infinte loops.

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2007-05-24  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	(rtsp_connection_send), (rtsp_connection_receive):
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
	Fix for new API.

	* gst/rtsp/rtspconnection.c: (add_auth_header),
	Only add authorisation and session headers when sending messages.

	* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_unset), (rtsp_message_add_header),
	(rtsp_message_remove_header), (rtsp_message_get_header),
	(rtsp_message_append_headers), (dump_key_value),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Add support for multiple headers of the same type by storing the parsed
	headers in a GArray instaed of a hashtable.

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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
	Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
	safer shutdown.

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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Added signal for backwards compat.

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2007-05-21  Sebastian Dröge  <slomo@circular-chaos.org>
	
	Patch by: René Stadler <mail at renestadler dot de>

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Use audioconvert for converting from non-native endianness floats
	in auparse instead of doing it ourself. Fixes #424527.
	This needs the audioconvert from plugins-base CVS.
	
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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_flush):
	Fix enum registration.

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2007-05-21  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
	(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
	(gst_rtp_h263p_pay_flush):
	* gst/rtp/gstrtph263ppay.h:
	Add new fragmentation mode base on GOB headers. Fixes #438940.

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2007-05-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
	  Printf format fix.

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2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Don't crash when an unsupported transport error was returned by the
	server, just try to configure the next stream. Fixes #439255.

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2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Add TCP timeout property and use it for all TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_write), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	Make connect and writes cancelable and make them use the timeout.

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2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Refactor timeout handling.
	Also send keep-alive when dealing with TCP transport.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_free), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	* gst/rtsp/rtspconnection.h:
	Use a timer to handle the session timeouts, add some methods to deal
	with timeouts.

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2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Ignore streams that fail the setup command, we will retry with a
	different transport later on.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_configure_stream):
	Fix encoding name case.

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2007-05-16  Edward Hervey  <edward@fluendo.com>

	* ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
	Fix build on macosx.

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2007-05-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
	Replace direct comparison of a string with the string literal "" with
	a comparison of the first character with '\0'. Fixes #438926.

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2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c (gst_break_my_data_init):
	  One more try. This should be the proper fix now.

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2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c:
	  Ooops, no // comments please.

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	* gst/debug/breakmydata.c: (gst_break_my_data_class_init),
	(gst_break_my_data_init):
	  Fix gst_buffer_is_writable() assertion.

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2007-05-14  David Schleef  <ds@schleef.org>

	* sys/v4l2/gstv4l2src.c: Add support for Bayer images as
	  video/x-raw-bayer.  Fixes #314160.

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	* gst/rtp/gstrtptheoradepay.c: (decode_base64),
	(gst_rtp_theora_depay_parse_configuration):
	* gst/rtp/gstrtptheorapay.c: (encode_base64),
	(gst_rtp_theora_pay_finish_headers),
	(gst_rtp_theora_pay_handle_buffer):
	Update theora pay/depayloader in a similar to vorbis.

	* gst/rtp/gstrtpvorbisdepay.c:
	(gst_rtp_vorbis_depay_parse_configuration):
	Update docs.

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	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
	When we try to execute a method that is not supported by the server,
	don't error out but remove the method from the accepted methods so that
	we never try to perform this method again.

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	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	Remove annoying _dump_mem.

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	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
	Parse range correctly.

	* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	The baseurl now always has a '/' at the start.

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	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	Factor out caps configuration and configure more stuff such as the time
	ranges and speed/scale values.

	* gst/rtsp/rtsptransport.c:
	Add Copyright after non-trival fixes.

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	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
	(rtsp_message_get_header):
	* gst/rtsp/rtspmessage.h:
	Make channel guint8 where possible.
	Make rtsp_message_init_data() take the channel as a guint8.

	* gst/rtsp/rtspdefs.c:
	Fixed a typo: Timout -> Timeout

	* gst/rtsp/rtspdefs.h:
	Make RTSP_CHECK() behave as a statement.

	* gst/rtsp/sdpmessage.c:
	Avoid a compiler warning in INIT_ARRAY().
	Fixes #437692.

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	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
	(rtsp_url_get_request_uri):
	* gst/rtsp/rtspurl.h:
	Add support for query parameters to RTSP URLs.

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2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
	(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	(rtsp_transport_parse), (rtsp_transport_as_text):
	* gst/rtsp/rtsptransport.h:
	Add validation to rtsp_transport_parse().
	Add rtsp_transport_as_text() to generate an RTSP header from an
	RTSPTransport.
	Change ssrc to guint (was a string) since that is what it is, even
	though it is sent as a hex string.
	Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
	incorrect, which can be seen when looking at the examples in the RFC).
	Fixes #437670.

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2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	Patch by: Eric Anholt

	* sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
	  gst_ximage_src_ximage_get):
	Use union of all damage between frames to make it faster.
	Fixes bug #342463.
	Also fix crasher when cursor is at bottom right of window.

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2007-05-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
	  streaming mode regression for file from #343837 with 'bext' chunk
	  before the 'fmt' chunk.

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2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.h:
	Preliminary seek support.
	Activate internal pads so that we can receive events on them.
	Don't try to parse a range string when it's NULL.

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2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with new RTP variables that will be used for
	synchronisation.

	* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
	(gst_rtp_vorbis_depay_parse_configuration),
	(gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (encode_base64),
	(gst_rtp_vorbis_pay_finish_headers),
	(gst_rtp_vorbis_pay_handle_buffer):
	Update vorbis pay and depayloader to draft-04.

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2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	UDP MCAST is actually the default for RTP/AVP.
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2007-05-13  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 can build 
	in_data += (filter->width / 8).
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2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
	  gst_ximage_src_ximage_get):
	* sys/ximage/gstximagesrc.h (last_ximage):
	When using Damage actually keep the last frame, and not assume
	that the buffer we get already has the last frame on it.
	Copy the cursor over if we specify a non-zero start x and
	start y.

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2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Make UDP the default transport when not specified.

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2007-05-09  David Schleef  <ds@schleef.org>

	* gst/level/gstlevel.c:
	  Revert last change.

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2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
	(gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 know the size of data
	pointed when moving the pointer.
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Move instructions after variables declaration.
	* win32/vs6/autogen.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update vs6 project files.

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2007-05-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
	* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
	(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
	(rtsp_range_free):
	* gst/rtsp/rtsprange.h:
	Add code to parse time ranges.
	Report DURATION on the stream when possible.

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2007-05-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
	(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
	(gst_videomixer_collected):
	  Fix strides calculation for AYUV (it's just width*4) (#436910).

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2007-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	Sync the GObject properties before each processing step to properly
	work with the controller.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state):
	Let more error state trickle down so that we can catch more error
	cases.
	Handle keep-alive a little smarter by selecting a method the server
	actually supports.
	Fix a race in UDP streaming shutdown.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
	Ignore errors when trying to use the keep-alive messages.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (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):
	Send RTCP messages back to the server over the TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Factor out and expose lowlevel _write and _read methods.
	Implement sending data messages to the server.

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2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
	(gst_multipart_mux_collected):
	Fix timestamps on outgoing buffers.

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2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c:
	(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Emit NEWSEGMENT events before pushing the first buffer.

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2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (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_push_event),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	Refactor transport configuration code.
	Create internal pads for TCP transport so that we can implement events
	and queries.
	Handle events and queries.
	Parse range from the SDP.
	Fix race in pause handler where the connection could still be flushing.

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2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Fix race when multiple udp sources post timeouts, just act on the first
	received timeout.
	Protect stream list with a recursive lock to fix some races.
	Flush connection when we need to do a reconnect or stop.
	Make state lock recursive.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close):
	Some small cleanups.

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2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	Only set DISCONT when there actually is a discont or when we just
	started.

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2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflac.c: (plugin_init):
	Call bindtextdomain() to get localized strings.

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2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Be a bit more clever when dealing with VBR files with FACT tags, we
	don't want to timestamp buffers in that case but the estimated BPS can
	be used for seeking.
	Only send close segment in the streaming thread.

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2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	Correctly post an error on the bus if something went wrong in the loop
	function. This fixes a few cases where the task was paused and nothing
	happened anymore.

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2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix compilation of deprecated test just because I'm too lazy to delete
	it.

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2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	Fix sending RTCP to the right place.
	Fix bug in reffing the wrong UDP element.
	Use new pad names for the session manager.
	Implement handling server requests in interleaved and UDP modes.
	Handle session keep-alive in UDP modes.
	Remove GCond for handling UDP timeouts.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive), (rtsp_connection_close):
	* gst/rtsp/rtspconnection.h:
	Store connection IP address for later.
	Add timeout args to all operations that might block forever.
	Parse session timeout.
	Only close sockets when not already closed.

	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Add timeout return value and error string.

	* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
	Add small comment.

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2007-05-01  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
	* gst/rtp/gstrtpmp4vpay.h:
	Handle NEWSEGMENT and FLUSH events. Fixes #434824.

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2007-04-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Remove v4l2src from docs, since it breaks the docs build, and the
	  plugin is only built if --enable-experimental is used anyway.

	* docs/plugins/Makefile.am:
	  Spaces => tab.

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2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (leave_multicast),
	(gst_multiudpsink_add), (gst_multiudpsink_remove):
	Add code to drop membership of a multicast group.

	* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
	(gst_udpsink_set_uri):
	Implement URI handler.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	Use URI handler to make udpsink instace.
	Improve code to configure port and destination.

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2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	Fix multicast detection.
	Don't try to join a multicast group if the address is not multicast.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
	Small debug improvement.

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2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message):
	Ignore ASYNC state messages from the udpsink, it's irrelevant for the
	parent.

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2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpilbcdepay.h:
	Fix mode property when specified as an arg.

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2007-04-26  Edward Hervey  <edward@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-osxaudio.xml:
	Add documentation for osxaudio plugin.

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2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect state changes with a lock.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(parse_line):
	* gst/rtsp/rtspconnection.h:
	Remove some unused stuff.

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2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Handle the case where there are exactly 0 bytes to read and the ioctl
	did not report an error. Fixes #433530.

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2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Apply DISCONT to buffers.
	Only apply timestamp to the first sample after a DISCONT, too many VBR
	files cause random jitter in the timestamps. Fixes #433119.

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2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property):
	* gst/rtsp/gstrtpdec.h:
	Add dummy latency property to be backwards compat with rtpbin.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Add latency property and configure in the session manager.
	Don't set invalid clock-base and seqnum-base on caps, some servers
	sometimes don't send them.

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2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
	(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
	  Double-check that RGB input caps are really RGBA caps (apparently
	  the core doesn't always catch it if those caps aren't a subset of
	  our template caps, also see #421543). Fixes #429319 in a way.
	  Also, don't leak the pad template in the transform_caps function.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/alphacolor.c: (setup_alphacolor),
	(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
	(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
	(GST_START_TEST), (alphacolor_suite):
	  Add some basic unit tests for alphacolor.

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2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  If we get a fatal flow return in the loop function, first post the
	  error message and only then send the EOS event downstream, otherwise
	  applications might get an eos message before the error message and
	  think everything was ok (related to #429319).

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2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	Read the channel byte as an unsigned byte.

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2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
	(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
	(gst_rtp_gsm_depay_setcaps):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
	(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
	(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
	(gst_ilbc_depay_get_property):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
	(gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
	(gst_rtp_pcmu_depay_setcaps):
	Make sure we configure the clock_rate in the baseclass in the setcaps
	function. Fixes #431282.

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2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_stream_free), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Parse server address from SDP.
	Hook up a udpsink to send RTCP back to the server.

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtsp/rtsptransport.h:
	Add some docs.

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2007-04-25  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Make header field check conditional. Fixes #433135

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot 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-alphacolor.xml:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  Add minimal docs blurb to alphacolor; split out headers into
	  separate header file for gtk-doc.

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2007-04-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c: (gst_progress_report_report):
	  Don't try to post NULL message (in case we can't query upstream
	  position or duration).

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2007-04-18  Michael Smith  <msmith@fluendo.com>

	* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
	(gst_cutter_get_caps):
	* gst/cutter/gstcutter.h:
	  Fix some of the most obvious bugs in cutter. Now doesn't leak
	  everything if input is silent.

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2007-04-18  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Wav apparently only supports width==GST_ROUND_UP(depth), everything
	else results in a invalid block align and invalid files.

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2007-04-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Snaik <snaik32 gmail com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
	  Add missing break statement for BOX_HORIZONTAL case.

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	Patch by: Vincent Torri <vtorri at univ-evry dot fr>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	Use correct format strings for integer types.

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	* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	(gst_wavparse_create_sourcepad):
	Use gst_riff_create_audio_template_caps () instead of the local caps.
	This makes updates of the local caps unecessary whenever libgstriff
	gets support for new formats.

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2007-04-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Brian Cameron  <brian.cameron at sun dot com>

	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/sunaudio/gstsunaudiosrc.h:
	  Fix and/or update copyright attributions (#430228).

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2007-04-13  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	Fix docs.

	* gst/rtsp/URLS:
	Add some more example urls.

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_chain_rtp):
	Better debugging.

	* gst/rtsp/gstrtspsrc.c: (request_pt_map),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_parse_rtpinfo):
	Remove unused code.

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2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Relax the audio/mpeg caps again and add FIXME: comment.

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	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	  More sanity check for the header fields. Fix type for 'rate' header
	  field.

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

	* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
	(gst_icydemux_unicodify):
	  If the metadata strings we get in the stream are not UTF-8, try to
	  interpret them according to the character encodings specified in the
	  GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
	  only fall back to locale/ISO-8859-1 if those aren't set or don't
	  work. Should fix #428901.

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	* gst/rtp/gstrtph264depay.c:
	Use the proper sync word for SPS and PPS.

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	* gst/rtp/Makefile.am:
	* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
	  fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
	* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
	  Add a simple hashing implementation that we can use to generate
	  a 24-bit ident value based on the codebooks for vorbis and theora.
	* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
	  gst_rtp_theora_pay_handle_buffer):
	* gst/rtp/gstrtpvorbisdepay.c
	  (gst_rtp_vorbis_depay_parse_configuration,
	  gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
	  gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
	  gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
	  Use the hashing function, ensuring that the same codebooks result
	  in the same ident and thus the same SDP description.
	  Various log fixes/changes.

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

	Patch by: jerry tan <jerry dot tan at sun dot com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
	application's responsibility to make sure it open the device once.
	Remove a careless error if AUDIODEV is set. Fixes #392620.

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

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
	* gst/rtsp/gstrtpdec.h:
	Make backward compat with rtpbin by adding the request-pt-map signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams):
	* gst/rtsp/gstrtspsrc.h:
	Implement request-pt-map signals instead of setting caps on the buffers
	for the session manager.

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2007-04-11  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudp.c: (plugin_init):
	Register GstNetBuffer in plugin_init so that the type can be used from
	multiple threads without races.

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2007-04-10  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	Fix depayloader clock_rate and some cleanups.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	* gst/rtp/gstrtph264depay.h:
	Don't push codec_data in the adapter because it might get flushed when
	we get a discont.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Handle multiple AU per packet.

	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
	(gst_rtp_sv3v_depay_plugin_init):
	Disable rank, this one does not work.
	Remove timestamping, base class does that.

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

	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	  limit caps to the formats we announce in the template

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
	  fix some crashers/asserts when dealing with broken files

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2007-04-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
	(gst_rtp_speex_depay_setcaps):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
	Fix some compiler warnings. Fixes #428182.

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

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
	(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_finalize),
	(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
	(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
	(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
	(create_rtcp), (gst_rtp_dec_request_new_pad),
	(gst_rtp_dec_release_pad):
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/gstrtsp.c: (plugin_init):
	Morph RTPDec into something compatible with RTPBin as a fallback.
	Various other style fixes.

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
	(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
	(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
	(new_session_pad), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Implement RTPBin session manager handling.
	Don't try to add empty properties to caps.
	Implement fallback session manager, handling.
	Don't combine errors from RTCP streams, just ignore them.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
	* gst/rtsp/rtsptransport.h:
	Implement fallback session manager.
	Make RTPBin the default one when available.

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

	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
	This element is ready to be autoplugged.

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2007-04-05  Julien MOUTTE  <julien@moutte.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	Don't leave the offsets defined by upstream element on the
	compressed data buffer we are pushing downstream. Make them
	GST_BUFFER_OFFSET_NONE.

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

	* gst/avi/README:
	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
	  Don't abort on out-of-memory. Use stream-nr as unsigned integer only.

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

	* gst/smpte/barboxwipes.c:
	Fix error as spotted by Snaik <snaik32 at gmail dot com>

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

	* gst/wavparse/gstwavparse.c:
	Support audio/x-raw-float in wav files. This only works with
	plugins-base CVS, using an older version doesn't have any
	disadvantages though.

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

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Revert last change as we don't want plugins-good to depend on
	plugins-base CVS now.

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

	* configure.ac:
	Require gst-plugins-base CVS for audioconvert with non-native
	float support and width/depth fix in libgstriff.

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Don't swap the floats ourself if they're not in native endianness.
	Instead let audioconvert handle this. Fixes #339838.

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

	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
	(gst_rtp_h263p_depay_change_state):
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	(gst_rtp_h264_depay_change_state):
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Flush adapter on disconts.

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

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
	Use more efficient adapter and rtpbuffer methods when possible.

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

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps):
	Correctly handle width!=depth input.
	* gst/wavparse/gstwavparse.c:
	Already export in the caps that width==8 uses unsigned samples and
	everything else uses signed samples.

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

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

	* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
	(gst_dynudpsink_init), (gst_dynudpsink_set_property),
	(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
	(gst_dynudpsink_close):
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Rework the socket allocation a bit based on the sockfd argument so that
	it becomes usable.
	Add a closefd property to instruct the udp elements to close the custom
	file descriptors when going to READY. Fixes #423304.
	API:GstUDPSrc::closefd property
	API:GstDynUDPSink::closefd property

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

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

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
	(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
	(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
	(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
	(gst_rtp_h264_pay_plugin_init):
	* gst/rtp/gstrtph264pay.h:
	Added H264 payloader. Fixes #423782.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	Small fixes.

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

	* gst/wavparse/gstwavparse.c:
	Actually support depths from 1 to 32, not only 8 to 32.

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

	* gst/wavparse/gstwavparse.c:
	Add support for wav files containing audio/x-raw-int with random
	depths between 1 and 32 bits.

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

	Based on patch by: Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
	(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
	(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
	(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
	(gst_rtp_mp4a_depay_get_property),
	(gst_rtp_mp4a_depay_change_state),
	(gst_rtp_mp4a_depay_plugin_init):
	* gst/rtp/gstrtpmp4adepay.h:
	Added MP4A-LATM depayloader. Fixes #417792.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	Fixup depayloader, setting codec_data, using more efficient adaptor and
	rtpbuffer handling.

	* gst/rtsp/URLS:
	Add url to test above.

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2007-03-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
	(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Handle default clock-rates for static payload types, rearrange stuff so
	that the rtpmap field in the sdp can override the defaults.
	Parse RTP-Info field to get the seqnum and timebase fields that should
	go in the caps.
	Delay configuring caps after we got the RTP-Info from the PLAY reply from
	the server. 

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

	Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Accept complex pipeline descriptions as an audio profile instead of just
	a single element. Fixes #420658.

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

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
	  Rename registered type in preparation of GstTagDemux moving to
	  -base at some point in the future.

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2007-03-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Streaming mode fixes: don't unref buffer we don't own any longer;
	  remove bogus adapter flush. Fixes #419338.

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

	* REQUIREMENTS: Change the format to key/value, add a bunch of
	  information, remove a bunch of requirements that are for
	  other GStreamer packages.

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

	* REQUIREMENTS: Fix a few things.  This file really needs a
	good once-over.

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2007-03-15  Edward Hervey  <edward@fluendo.com>

	* sys/Makefile.am:
	Don't forget to distribute the sys/osxaudio/ directory.

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2007-03-15  Edward Hervey  <edward@fluendo.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudiosink.c:
	(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
	(gst_osx_audio_sink_getcaps),
	(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
	* sys/osxaudio/gstosxaudiosrc.c:
	(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
	(gst_osx_audio_src_create_ringbuffer):
	* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
	(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
	(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
	(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
	* sys/osxaudio/gstosxringbuffer.h:
	Activate osxaudio in gst-plugins-good with proper build setup.
	Add inlined documentation.
	Fix debug statements
	Fix ringbuffer when pausing.
	Fixes #323471

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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	Ported mulaw and alaw payloaders to use new base class

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

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

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2007-03-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix string replace error (AG_AG_GST_* => AG_GST_*).

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2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking,
	  and SEEK_CUR+SEEK_END here as well.

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2007-03-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking, 
	  and SEEK_CUR+SEEK_END.

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2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
	  the image format a variable-length NUL-terminated string; in
	  versions before that the image format is a fixed-length string of
	  3 characters (see #348644 for a sample tag).
	  Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.

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

	* win32/MANIFEST:
	Add new project files to MANIFEST.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.
	
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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
	(gst_avi_demux_parse_index):
	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Printf format fixes; also add some missing quotes in translated
	  strings. Fixes #416728 and #416727.

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
	  Tim and I can't think of any reason the child audio sink needs to 
	  be set back to NULL after successfully determining that it can 
	  reach READY - it gets immediately set back to READY by the caller
	  anyway, causing an unnecessary close/open of any audio devices
	  involved.

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

	* po/LINGUAS:
	* po/ja.po:
	  Add ja.po file from #377306.

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

	* sys/sunaudio/gstsunaudio.c: (plugin_init):
	* sys/sunaudio/gstsunaudiomixertrack.c:
	(gst_sunaudiomixer_track_new):
	  Actually translate sunaudio mixer track labels instead of just
	  marking the strings as translatable (#377306); clean up weird
	  label string mapping code that serves no apparent purpose. Also
	  set the 'untranslated-label' property when creating mixer tracks
	  if the GstMixerTrack base class supports this.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/sunaudio.c: (GST_START_TEST),
	(sunaudio_suite):
	  Very minimalistic unit test for sunaudiomixer element (compiles, but not
	  actually tested on a system where sunaudiomixer is available).

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Re-enable the states test and see if it works on the buildbots.

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
	(gst_dvdec_src_negotiate), (gst_dvdec_chain),
	(gst_dvdec_change_state):
	* ext/dv/gstdvdec.h:
	Infer pixel-aspect-ratio from the video frame format if it isn't
	provided by the container, as happens when playing DV from AVI
	or Quicktime containers.

	Patch by: Wim Taymans <wim@fluendo.com>
	Fixes #380944

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	When activated, remove the udpsrc timeout, we have dataflow and timeouts
	will later be handled by the jitterbuffer.

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

	* ext/taglib/gstid3v2mux.cc:
	Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.
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	Patch by: Alex Lancaster <alexl at users sourceforge net>
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2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix stream position reporting after a seek. Fixes #416445.

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

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_chain):
	Make avidemux accept optional header chunks in any order.
	Fixes #415446.

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

	* tests/check/Makefile.am:
	Disable the states check until the remaining Valgrind errors
	are fixed or suppressed.

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

	* tests/check/elements/.cvsignore:
	  Add audiodynamic check to .cvsignore

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

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_characteristics_get_type),
	(gst_audio_dynamic_mode_get_type),
	(gst_audio_dynamic_set_process_function),
	(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
	(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
	(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
	(gst_audio_dynamic_transform_hard_knee_compressor_int),
	(gst_audio_dynamic_transform_hard_knee_compressor_float),
	(gst_audio_dynamic_transform_soft_knee_compressor_int),
	(gst_audio_dynamic_transform_soft_knee_compressor_float),
	(gst_audio_dynamic_transform_hard_knee_expander_int),
	(gst_audio_dynamic_transform_hard_knee_expander_float),
	(gst_audio_dynamic_transform_soft_knee_expander_int),
	(gst_audio_dynamic_transform_soft_knee_expander_float),
	(gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new audiodynamic element which can act as a compressor or
	expander. Supported are hard-knee and soft-knee operation modes with
	user-specified ratio and threshold.
	Attack and release parameters are not yet implemented but will follow.
	* 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-audiofx.xml:
	Integrate audiodynamic into the docs.
	* tests/check/Makefile.am:
	* tests/check/elements/audiodynamic.c: (setup_dynamic),
	(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
	Add unit test for audiodynamic.

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

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
	Free handles that we allocated when exiting via the error paths.

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2007-03-07  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_class_init),
	(gst_level_set_caps), (gst_level_start), (gst_level_event),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	  Resolve message timestamps against the playback segment.

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2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
	(gst_id3demux_sink_activate):
	  Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
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	  caps passed to it (previously one code path assumed it took ownership
	  while another one assumed it didn't, while in fact it sometimes did and
	  sometimes didn't ...).
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	* configure.ac:
	* tests/files/Makefile.am:
	* tests/files/id3-407349-1.tag:
	* tests/files/id3-407349-2.tag:
	  Add directory where data for unit tests can be stored.

	* tests/Makefile.am:
	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
	(read_tags_from_file), (run_check_for_file),
	(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
	  Add unit test for id3demux, and in particular for bug #407349. Only
	  testing pull-mode for now; push mode doesn't work yet because the test
	  files are smaller than ID3_TYPE_FIND_MIN_SIZE.

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2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Add missing backslash at end of line.

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

	Trigger rebuild.

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2007-03-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	(parse_obsolete_tdat_frame):
	  Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
	  the four-digit number will be interpreted as a year, whereas it is
	  month and day in DDMM format. Instead, parse TDAT frames and fix up
	  the date in the GST_TAG_DATE tag later if we also extracted a year.
	  Fixes #407349.

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

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid):
	Fix up the dispose logic so it doesn't leak, and fix setting of 
	the child state so that we don't set a child to our current state 
	just as we are changing it to something else.

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

	* gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
	(gst_goom_chain):
	* gst/goom/gstgoom.h:
	Document, fix and improve goom adapter behaviour.
	Fixes #407006.

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

	* ext/esd/esdsink.c: (gst_esdsink_open):
	Unref static pad template after using it.

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

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid):
	Fix up the reference counting of the child elements.

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2007-03-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
	Fix encoding-name case.

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2007-03-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
	(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
	(gst_rtp_speex_depay_process):
	* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
	(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
	(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
	(gst_rtp_speex_pay_change_state):
	* gst/rtp/gstrtpspeexpay.h:
	Fix speex (de)payloader. Fixes #358040.

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

	* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
	(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
	Install fakesink in NULL by fixing some broken logic. This obviates
	the need to manually set _IS_SINK.
	Add some comments and remove a little cruft while I'm at it.

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	* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
	Mark us as a sink when we have no fakesink in NULL. Fixes #414887.

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2007-03-04  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Update.

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2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Gah! Also disable gconfvideosink from the tests, otherwise
	it will instantiate autovideosink, and dfbvideosink and
	leak on the buildbots.

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2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
	(gst_cdio_cdda_src_finalize):
	Make sure we always destroy our libcdio handle.

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2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable autovideosink so the buildbots don't barf over memory
	leaked in the directfb sink.

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	* sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose):
	Chain up in dispose

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	* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	(gst_multipart_find_pad_by_mime):
	Use gst_pad_new_from_static_template instead of
	static_pad_template_get+pad_new.

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	* sys/ximage/gstximagesrc.c: (gst_ximage_src_create):
	Catch the case where no clock has been set.

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2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
	(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
	* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
	(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
	(gst_gconf_audio_src_finalize), (do_toggle_element):
	* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
	(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
	(do_toggle_element):
	* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
	(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
	(gst_gconf_video_src_finalize), (do_toggle_element):
	* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
	(gst_switch_sink_reset), (gst_switch_sink_set_child):
	* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
	* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
	* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	(gst_shout2send_init), (gst_shout2send_finalize):
	* gst/debug/testplugin.c: (gst_test_class_init),
	(gst_test_finalize):
	* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
	(gst_flxdec_dispose):
	* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize):
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
	* gst/rtsp/rtspextwms.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
	(gst_smpte_finalize):
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
	* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
	(gst_udpsink_finalize):
	* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
	(gst_wavparse_sink_activate):
	* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
	* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
	(gst_oss_src_finalize):
	* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_finalize):
	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):

	Fix a bunch of leaks shown by the newly-added states test.

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2007-03-04  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/dv/gstdvdec.c: (gst_dvdec_init):
	Use gst_pad_new_from_static_template instead of 
	static_pad_template_get+pad_new.

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

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

	* ext/libcaca/Makefile.am:
	* gst/debug/Makefile.am:
	  Don't mix tabs and spaces (#414168).

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2007-03-02  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/generic/.cvsignore:
	  Ignore files to please buildbot.

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2007-03-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Unbreak my previous commit (swapped nominator & denominator). Tim,
	  thanks for spotting.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
	(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
	(gst_cdio_cdda_src_finalize):
	Small code cleanups.
	Don't use pad_alloc as the base class cannot deal with the error codes.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create):
	Fix doc.

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	Patch by: René Stadler <mail@renestadler.de>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Handle rounding better to not drop last sample frame. Fixes #356692

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

	* tests/check/Makefile.am:
	Disable cacasink from the states check too - it also calls exit(1)
	on us when it can't find a terminal to talk to.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property):
	* gst/udp/gstudpsrc.h:
	Add support to strip proprietary headers. Fixes #350296.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	Fix compilation.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init),
	(gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process),
	(gst_rtp_mp2t_depay_set_property),
	(gst_rtp_mp2t_depay_get_property):
	* gst/rtp/gstrtpmp2tdepay.h:
	Add support to strip off proprietary headers. Fixes #350278.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	* ext/hal/hal.c:
	Fix compilation.

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2007-03-02  Wim Taymans  <wim@fluendo.com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init),
	(gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property),
	(gst_sunaudiosrc_open):
	* sys/sunaudio/gstsunaudiosrc.h:
	Remove device-name from GstSunAudioSrc. Fixes #412597.

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2007-03-01  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/hal/gsthalaudiosink.c: (do_toggle_element):
	* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	Having NULL as UDI previously selected the default sink/src. Change
	this back but mention it in the debug output.
	* ext/hal/hal.c: (gst_hal_get_alsa_element),
	(gst_hal_get_oss_element), (gst_hal_get_string),
	(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
	(gst_hal_get_audio_src):
	* ext/hal/hal.h:
	Refactor a bit, check all error conditions, greatly improve debugging
	and fix some possible memory leaks. Also implement OSS support
	and allow specifying an UDI that points to a real device. For this the
	child device which supports ALSA (preferred) or OSS is used.
	As a side effect this makes it impossible now to get a alsasink in
	halaudiosrc and a alsasrc in halaudiosink.

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	* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
	(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
	Errors from the udp sources are not fatal unless all of them are in
	error.

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	* tests/check/Makefile.am:
	Disable aasink in the states test. I suspect this is the element that
	is calling exit(1) when it can't proceed.

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	* tests/check/Makefile.am:
	Draw plugins in from the build tree sys/ dir, rather than picking
	up the already installed versions.

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	* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
	Error out correctly when getting xcontext fails.

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	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
	Make state change to PAUSED NO_PREROLL because that's what it will be in
	the future and rtspsrc relies on it.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_change_state):
	Don't error out when we don't get an error from the state change
	function.

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	* ext/hal/gsthalaudiosink.c: (do_toggle_element):
	* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
	  Check if the device UDI is set before trying to query HAL
	  about it and give a useful error message if it wasn't set.
	* ext/hal/hal.c: (gst_hal_get_string):
	  Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
	  gives an assertion failure in D-Bus when running with
	  DBUS_FATAL_WARNINGS=1.

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	* configure.ac:
	  Convert to new AG_GST style.

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	* tests/check/Makefile.am:
	* tests/check/generic/states.c: (GST_START_TEST), (states_suite):
	  add test for states

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	* tests/check/elements/.cvsignore:
	Add new videofilter check to .cvsignore.

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	* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop), (gst_avi_demux_chain):
	Fix combined flow return. Fixes #412608.

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	* gst/videofilter/Makefile.am:
	Dist header..

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	* gst/videofilter/gstgamma.h:
	Add header too.

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	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/videofilter/Makefile.am:
	* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
	(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
	(gst_gamma_get_property), (gst_gamma_calculate_tables),
	(oil_tablelookup_u8), (gst_gamma_set_caps),
	(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
	Port gamma filter to 0.10. Fixes #412704.

	* tests/check/Makefile.am:
	* tests/check/elements/videofilter.c: (setup_filter),
	(cleanup_filter), (check_filter), (GST_START_TEST),
	(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
	Add unit tests for videofilters.

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	* gst/rtsp/URLS:
	Add another interesting test url.

	* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
	Don't allow getting header fields from data packets.

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	* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
	(gst_shout2send_init), (gst_shout2send_start),
	(gst_shout2send_set_property), (gst_shout2send_get_property):
	* ext/shout2/gstshout2.h:
	  Add a property for username.

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	* sys/osxaudio: Add Pioneers of the inevitable to the copyright list

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	* gst/rtsp/Makefile.am:
	Fix make check too.

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	* gst/rtsp/base64.c: (util_base64_encode):
	* gst/rtsp/base64.h:
	Commit missing files for base64 encoding.

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

	* configure.ac:
	* ext/annodex/Makefile.am:
	* ext/jpeg/Makefile.am:
	* ext/speex/Makefile.am:
	* gst/alpha/Makefile.am:
	* gst/cutter/Makefile.am:
	* gst/debug/Makefile.am:
	* gst/effectv/Makefile.am:
	* gst/goom/Makefile.am:
	* gst/level/Makefile.am:
	* gst/smpte/Makefile.am:
	* gst/videofilter/Makefile.am:
	  Fix build with LDFLAGS='-Wl,-z,defs' (#410997)

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2007-02-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/rtspconnection.c: (append_auth_header),
	(rtsp_connection_send), (rtsp_connection_set_auth):
	g_base64_encode is a GLib 2.12 function. Use an equivalent taken
	from icecast to replace it. Relicensed from GPL courtesy of Mike
	Smith.

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	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
	(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(append_auth_header), (rtsp_connection_send),
	(rtsp_connection_free), (rtsp_connection_set_auth):
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	* gst/rtsp/rtspurl.h:

	Implement simple Basic Authentication support so that urls like
	rtsp://user:pass@hostname/rtspstream work on hosts that require
	authentication.

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	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/v4l2_calls.c:
	Fix segfault when oppening a radio device.
	
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2007-02-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_set_caps),
	(gst_level_transform_ip):
	* sys/v4l2/README:
	* tests/check/elements/level.c: (GST_START_TEST):
	  Fix level for multi-channel case.

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

	* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	  Use function pointer for process function and add process functions
	  for float audio.

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2007-02-19  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init):
	  Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
	  fixes #407369

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2007-02-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
	(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
	(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
	(gst_rtp_mp2t_pay_plugin_init):
	* gst/rtp/gstrtpmp2tpay.h:
	Added simple mpeg transport stream payloader.

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2007-02-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add example H264 rtsp url.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	Don't convert values to lowercase or we might mess up base64 encoded
	properties.

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2007-02-16  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Fix case of string params.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	Fix depayloader, support more packet types.
	Add sync codes to make sure the packetizer can do its job.

	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	Fix caps case again.

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

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	Set right caps on output buffers.

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2007-02-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/sdpmessage.c: (sdp_parse_line):
	As spotted by: Peter Kjellerstedt  <pkj at axis com>:
	Clear stack allocated SDPMedia struct before calling _init() on it.
	Clarify this in the docs as well.

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2007-02-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
	(do_change_child):
	Don't reset the profile when going switching states, as it makes
	the element non-reusable.

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2007-02-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
	(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
	(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
	(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
	(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
	(sdp_parse_line):
	* gst/rtsp/sdpmessage.h:
	Based on patch by: jp.liu <jp_liu at astrocom dot cn>
	Fix memory management of SDP messages. Fixes #407793.

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2007-02-14  Stefan Kost  <ensonic@users.sf.net>

	Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>

	* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
	Allow muxing video/x-h264 (was already in the caps). Fixes #407780.

2007-02-14  Wim Taymans  <wim@fluendo.com>
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	Patch by: jp.liu <jp_liu at astrocom dot cn>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of password field in url. Fixes #407797.

2185
2007-02-14  Wim Taymans  <wim@fluendo.com>
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	* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
	(gst_wavparse_reset), (gst_wavparse_init),
	(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
	(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
	(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	(gst_wavparse_loop), (gst_wavparse_chain),
	(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
	(plugin_init):
	* gst/wavparse/gstwavparse.h:
	Update docs.
	Use boilerplate.
	Various code cleanups.
	When the bitrate is not known (bps == 0 or compressed formats) let
	downstream element guestimate the duration and position and don't
	generate timestamps or durations. Fixes #405213.
	Fix EOS and ERROR conditions in chain mode, we just need to forward the
	error flowreturn upstream.

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2007-02-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/Makefile.am:
	* ext/gconf/gconf.c: (gst_gconf_get_string),
	(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
	(gst_gconf_render_bin_with_default):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
	(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
	(gst_gconf_audio_sink_dispose), (do_change_child),
	(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
	(cb_change_child), (gst_gconf_audio_sink_change_state):
	* ext/gconf/gstgconfaudiosink.h:
	* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
	(gst_switch_sink_class_init), (gst_switch_sink_reset),
	(gst_switch_sink_init), (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
	(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
	(gst_switch_sink_get_property), (gst_switch_sink_change_state):
	* ext/gconf/gstswitchsink.h:
	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
	(gst_auto_audio_sink_detect):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
	(gst_auto_video_sink_detect):
	Re-factor the gconfaudiosink into a "GstSwitchSink" base class
	and a child that implements the GConf key monitoring. The end goal of
	this is an audio sink that can be changed on the fly, but at the 
	moment it still only changes on the next READY transition.

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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif

2252
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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/monoscope/Makefile.am:
	* gst/monoscope/gstmonoscope.c:
	  Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
	  (but no LIBS, since we only use defines from the headers).

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
	(gst_wavparse_stream_data):
	  Fix massive memory leak when operating in streaming mode due to
	  GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
	  Fixes #407057.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	* gst/avi/gstavidemux.h:
	  Save some memory (8%) by repacking the index entry structure (more to
	  come). Add more FIXMEs to questionable parts.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
	(gst_v4l2src_get_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init):
	  More FIXME comments and messaging changes.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	(gst_goom_change_state):
	* gst/goom/gstgoom.h:
	  Improved docs and use GST_DEBUG_FUNCPTR.

	* gst/level/gstlevel.c: (gst_level_class_init):
	  Use GST_DEBUG_FUNCPTR.

	* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
	(gst_monoscope_chain), (gst_monoscope_change_state):
	  Improved docs source cleanups.

2315
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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/Makefile.am:
	* gst/debug/gstdebug.c: (plugin_init):
	* gst/debug/gstpushfilesrc.c:
	* gst/debug/gstpushfilesrc.h:
	  Add code for a pushfilesrc element that implements a pushfile:// URI
	  handler, to make debugging push-mode operation of demuxer/decoders
	  that support both easier in connection with seek/playbin/etc.
	  The element isn't registered at the moment.

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2007-02-11  Sébastien Moutte  <sebastien@moutte.net>

	* gst/avi/gstavimux.c:
	  Comment a #if 0 in caps template definition as VS6 seems to 
	do not support it.
	* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
	  Use gst_guint64_to_gdouble for conversion.
	* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
	  Move variables declaration before the first instruction.
	* gst/rtsp/rtspdefs.c:(rtsp_strresult):
	  Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
	  And don't include netdb.h for G_OS_WIN32
	* gst/rtsp/sdpmessage.c:(sdp_parse_line):
	  This initialization SDPMedia nmedia = {.media = NULL }; is not supported
	  by VS6 then use an other way to initialize SDPMedia structure.
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstdynudpnetutils.h:
	  Do not include <sys/time.h> for G_OS_WIN32
	* gst/udp/gstudpsrc.c:
	  Define socklen_t as int for G_OS_WIN32
	* win/common/config.h.in:
	  Undef HAVE_NETINET_IN_H
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstautogen.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstudp.dsp:
	  Add and update project files.
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	  Add a copy of udp enumtypes to win32/common as in core 
	  and base.
	
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2007-02-11  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Activate monoscope when building with --enable-experimental. Fix
	  --enable-external configure switch description.

	* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
	  Help gst-indent.

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2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  Explicitly cast result of pointer arithmetic to integer in order to
	  avoid compiler warnings on some 64-bit systems. Should fix #406018.

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2007-02-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c:
	  Some more docs.

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2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/inspect/plugin-rtp.xml:
	  Update for new elements.

	* gst/debug/progressreport.h:
	  Commit newly-created header file as well.

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2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* 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.hierarchy:
	* gst/debug/Makefile.am:
	* gst/debug/progressreport.c: (gst_progress_report_post_progress),
	(gst_progress_report_do_query), (gst_progress_report_report):
	  Make progressreport element post messages with the current progress
	  on the bus. Also add some basic docs for it.

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2007-01-30  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/hal/hal.c: (gst_hal_get_string):
	* ext/hal/hal.h:
	  Some small cleanups; deal with errors when parsing the HAL ALSA
	  capabilities a bit better.

2408
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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Let's try this again and use the right cast this time.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Add cast to avoid compiler warnings with older GLib versions
	  where the nick/name members in GEnumValue are not declared as
	  constant strings.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
	(gst_gconf_render_bin_from_key),
	(gst_gconf_get_default_audio_sink):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
	(do_toggle_element), (gst_gconf_audio_sink_set_property),
	(gst_gconf_audio_sink_get_property):
	  In gconfaudiosink, get the right key as the old key in