ChangeLog 316 KB
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2006-11-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  We require a -base more recent than 0.10.9, so it's safe to use
	  GST_TYPE_TAG_IMAGE_TYPE unconditionally now.

	* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
	  Use _newsegment_full() now that we depend on a recent enough core.

	* gst/wavparse/gstwavparse.c:
	  Remove cruft that we don't need any longer now that we depend on
	  a recent enough -base.

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2006-10-31  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
	(gst_rtpilbcpay_setcaps):
	Fix and activate ILBC pay and depayloaders. Fixes #368162.

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2006-10-31  Wim Taymans  <wim@fluendo.com>

	* ext/speex/gstspeexdec.c: (speex_dec_convert),
	(speex_dec_sink_event), (speex_dec_chain_parse_header):
	Some small cleanups, use _scale.

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2006-10-31  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	Use higher precision scale function.

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2006-10-30  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Michal Benes  <michal dot benes at itonis tv>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_decode_buffer):
	  Fix several issues with encoded/compressed/encrypted/signed tracks;
	  also, remove superfluous newline characters from some debug
	  statements. (#366155)

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2006-10-30  Wim Taymans  <wim@fluendo.com>

	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
	(gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
	(gst_smokedec_change_state):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
	(gst_smokeenc_init), (gst_smokeenc_finalize),
	(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
	(gst_smokeenc_resync), (gst_smokeenc_chain),
	(gst_smokeenc_set_property), (gst_smokeenc_get_property),
	(gst_smokeenc_change_state):
	Various cleanups, capsnego and leak fixes.

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2006-10-30  Wim Taymans  <wim@fluendo.com>

	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
	Fix videomixer so that it can handle any combination of framerates.
	Fixes #367221.

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2006-10-28  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_file_header),
	(gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix position query for audio. also fixes timestamps in streaming
	mode and bug #364958.
	Small cleanups.

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2006-10-27  Wim Taymans  <wim@fluendo.com>

	* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain):
	* ext/libpng/gstpngenc.h:
	Fix strides. Fixes #364856.
	Cleanup capsnego.
	Set caps on outgoing buffers.

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2006-10-18  Wim Taymans  <wim@fluendo.com>

	Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>

	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	(gst_rtp_pcma_pay_handle_buffer):
	* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
	Add static payload numbers in addition to the dynamic ones.
	Fixes #361639.

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2006-10-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/rtspurl.h:
	Reuse already existing enum for lower transport.
	Add rtspt and rtspu protocols.
	Send redirect to rtspt when udp times out.

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2006-10-18  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_data):
	Fix seeking some more, mostly for speed changes.

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2006-10-18  Tim-Philipp Müller  <tim at centricular dot net>

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	Patch by: Fredrik Persson  <frepe at bredband net>
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	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	  Fix _set_channel(): remove useless g_object_notify() for "channel"
	  property that doesn't exist any longer and therefore now also
	  useless redirect (#338818).

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2006-10-17  Wim Taymans  <wim@fluendo.com>

	* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
	Some drivers do not support unsetting the non-blocking flag once the
	device is opened. In those cases, close/open the device in
	non-blocking mode. Fixes #362673.

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2006-10-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  dear stefan, framespersecond is not frameperiod, reverting but adding
	  comment

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2006-10-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  Numerator is numerator and denominator is denominator. Say that aloud
	  5 times and retry after next beer.

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2006-10-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Josep Torra Valles  <josep at fluendo com>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/esd/esdsink.c: (gst_esdsink_write):
	* ext/flac/gstflacdec.c: (gst_flac_dec_length),
	(gst_flac_dec_read_seekable), (gst_flac_dec_chain),
	(gst_flac_dec_send_newsegment):
	* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
	(gst_flac_enc_tell_callback):
	* ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
	(smokecodec_parse_header), (smokecodec_decode):
	* gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
	* gst/debug/efence.c: (gst_fenced_buffer_alloc):
	* gst/goom/Makefile.am:
	* gst/goom/gstgoom.c:
	* gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
	* sys/sunaudio/gstsunaudiomixertrack.h:
	  Fix a bunch of problems discovered by the Forte compiler, mostly type
	  mixups and pointer arithmetics with void pointers. Fixes #362603.

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2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeex.c: (plugin_init):
	* ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
	(gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
	(gst_speex_enc_class_init), (gst_speex_enc_finalize),
	(gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
	(gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
	(gst_speex_enc_src_query), (gst_speex_enc_sink_query),
	(gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
	(gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
	(gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
	(gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
	(gst_speex_enc_chain), (gst_speex_enc_get_property),
	(gst_speex_enc_set_property), (gst_speex_enc_change_state):
	* ext/speex/gstspeexenc.h:
	  Miscellaneous clean-ups, among other things: speexenc => enc to
	  enhance code readability; change speexenc => speex_enc; in chain
	  function unref input buffer in case of error; take reference in
	  event function; use boilerplate macro; use gst_pad_query_peer_*
	  convenience functions.

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2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
	(gst_speexenc_set_last_msg), (gst_speexenc_setup),
	(gst_speexenc_set_header_on_caps):
	  Fix some mem leaks.

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2006-10-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some other URL.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
	(gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Work on fallback to TCP connection when the UDP socket times out.
	Handler server requests, just reply with OK for now.

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Added some more Real extension headers.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of urls with a ':' that is not part of the hostname:port
	part of the url.

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2006-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
	* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Activate pad before adding it to the already-running element.

	* tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Activate newly-created pad too.

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2006-10-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Sebastien Cote <sebas642 at yahoo dot ca>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
	(gst_udpsrc_start):
	Fix some leaks in caps and uris. Fixes #361252.

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2006-10-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).

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2006-10-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	Fix build as per the patch in #338818 comment 36.

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2006-10-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
	  Activate pads before adding them to the source.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
	* gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
	Activate pads before adding.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	(gst_multipart_find_pad_by_mime):
	Activate pads before adding.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	BOILERPLATE sets parent_class for us.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_alloc_udp_ports),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Rework how the transport string is constructed, try to share channels
	and udp ports.
	Make most of the stuff less dependant on RTP as we are also going to use
	it for RDT.
	Add support for transport specific session managers.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
	Implement _flush().

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Add generic error return code.

	* gst/rtsp/rtspext.h:
	Add support for pluggable tranport strings.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
	(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	Detect WMServer and activate the extension.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
	(rtsp_transport_get_manager), (rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Added methods to get mime/manager for certain transports.

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2006-10-05  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c:
	(gst_cairo_time_overlay_update_font_height):
	* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	* ext/libpng/gstpngdec.c: (user_endrow_callback):
	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_data):
	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	* gst/debug/efence.c: (gst_efence_buffer_alloc),
	(gst_fenced_buffer_copy):
	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_handle_message):
	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	  Printf format fixes.

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2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	Dist new .h file too.

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2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
	(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Factor out extension in separate module.
	Fix getcaps to filter against the padtemplate.
	Use Content-Base if the server gives one.
	Rework the transport parsing a bit for future extensions.
	Added some Real Header field definitions.

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2006-10-04  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  added v4l2 stubs
	* gst-plugins-good.spec.in:
	  add v4l2

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2006-10-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  Extract disc/album/medium number and count and try harder
	  to extract track number/count.

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2006-10-03  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* sys/Makefile.am:
	  add build stuff for v4l2, needs --enable-experimental until
	  the last bits are resolved

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2006-09-29  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Disable autodetect test temporarily, so that the build bots
	  update -bad and the ranks of unreliable video sinks in there.

	* tests/check/elements/autodetect.c: (GST_START_TEST):
	  Skip test if no usable videosink is found.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add some more URLs.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_finalize),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add timeout property to control UDP timeouts.
	Fix error messages.
	Also start a loop function when operating in UDP mode so that we can
	do some more stuff async.
	Handle element messages from udpsrc to detect timeouts. If a timeout
	happens we currently generate an error.
	API: rtspsrc::timeout property.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create):
	Really implement the timeout in microseconds and not milliseconds.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Added property to post a message on timeout.
	Updated docs.
	When restarting the select, initialize the fdsets again.
	Init control sockets so we don't accidentally close a random socket.
	API: GstUDPSrc::timeout property

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
	Fix flag registration.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	Reading 0 also means 'no more commands'

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2006-09-29  Wim Taymans  <wim@fluendo.com>

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

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Fix possible infinite loop when shutting down, a read can also return
	0 to indicate no more messages are available. Fixes #358156.

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2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
	(gst_auto_audio_sink_find_best):
	* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
	Small cleanups.
	don't try to set "sync" property when it is not available.

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/alpha/gstalpha.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	* gst/videomixer/videomixer.c:
	  Include stdlib.h in some more places, makes things compile
	  with uClibc and -Werror (#357592).

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/jpeg/gstjpegdec.c:
	  Set minimum height to 8 (from 16), our code should handle
	  that fine. Some of the buttons on the apple trailer site
	  are apparently only 15 pixels high (see #357470).

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_open):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Improve error reporting.

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
	(gst_rtp_mp2t_depay_plugin_init):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
	Fix klass typos.
	Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.

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2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Need  -base CVS for gst_base_rtp_depayload_push_ts().

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
	Don't check for a tag that is never there and check if we read the
	correct tag. Fixes seeking again.
	We must post an error when all pads are unlinked.

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	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	(gst_rtp_vorbis_pay_reset_packet),
	(gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
	(gst_rtp_vorbis_pay_handle_buffer):
	More fixage, set endoder-params correctly in the payloader.

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2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_base_init):
	  Make static pad templates static to appease valgrind's leak
	  detector.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/autodetect.c: (GST_START_TEST),
	(autodetect_suite):
	  Add simple test for the ghostpad lockup on shutdown fixed in core
	  CVS (audio bit disabled because it would need dozens of alsa
	  suppressions and I'm too lazy to add those now).

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
	Small cleanups.

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
	(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
	(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
	(gst_rtp_vorbis_depay_process),
	(gst_rtp_vorbis_depay_set_property),
	(gst_rtp_vorbis_depay_get_property),
	(gst_rtp_vorbis_depay_change_state),
	(gst_rtp_vorbis_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
	(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
	(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet),
	(gst_rtp_vorbis_pay_append_buffer),
	(gst_rtp_vorbis_pay_handle_buffer),
	(gst_rtp_vorbis_pay_plugin_init):
	* gst/rtp/gstrtpvorbispay.h:
	Add experimental vorbis pay and depayloaders.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
	Fix profile-level-id parsing and setup.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/README:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	Update README, simple cleanup.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with some examples.

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
	(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
	(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
	(gst_rtp_mp4g_pay_setcaps):
	* gst/rtp/gstrtpmp4gpay.h:
	Make optional RTP parameters of type STRING, as required by the
	application/x-rtp caps specification.

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	* gst/rtp/gstrtph263pdepay.c:
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	* gst/rtp/gstrtph263ppay.c:
	Correctly calculate size of each H263+ RTP buffer taking into account MTU and
	RTP header.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	And makefile too.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
	(gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
	(decode_base64), (gst_rtp_asf_depay_setcaps),
	(gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
	(gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
	(gst_rtp_asf_depay_plugin_init):
	* gst/rtp/gstrtpasfdepay.h:
	Added preliminary ASF depayloader.

	* gst/rtp/gstrtph264depay.c: (decode_base64):
	Fix base64 decoding.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some test URLS.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_loop), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	When creating streams, give access to the complete SDP.
	Fix some leaks.
	Collect and merge global stream properties in stream caps.
	Preliminary support for WMServer.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Make connection interruptable.
	Refactor to make it reconnectable.
	Don't fail on short reads when reading data packets.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
	(rtsp_url_get_port):
	* gst/rtsp/rtspurl.h:
	Add methods for getting/setting the port.

	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_message_get_attribute_val), (sdp_media_get_attribute),
	(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
	(sdp_media_get_format), (sdp_parse_line),
	(sdp_message_parse_buffer):
	Fix headers. 
	Add methods for getting multiple attributes with the same name.
	Increase buffer size when parsing.
	Fix parsing of a=foo fields.

	* gst/rtsp/test.c: (main):
	Update to new connection API.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtsp.h:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Dual licensed under MIT and LGPL now.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Reorganize stream parsing and creation.
	Detect container formats in interleaved mode.
	Keep more state about the streams.
	Assume a server also supports PLAY if it does not say.
	Add unicast and interleaved properties to TCP transport requests to make
	some servers happy (WMServer).

	* gst/rtsp/sdpmessage.h:
	Add some defines for the standard Bandwidth types.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix build.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c:
	Add ms-gsm to the src template.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Small cleanups, added documentation.
	Try to clean up the requests and responses.
	Refactor parsing the supported methods.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
	(rtsp_connection_create), (rtsp_connection_send),
	(parse_response_status), (parse_request_line),
	(rtsp_connection_receive), (rtsp_connection_close),
	(rtsp_connection_free):
	* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
	(rtsp_transport_init), (rtsp_transport_parse),
	(rtsp_transport_free):
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
	(sdp_message_clean), (sdp_message_free), (sdp_media_new),
	(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
	Use g_return_val some more.

	* gst/rtsp/rtspdefs.h:
	Add more enum values to track initial states.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
	(rtsp_message_init_request), (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free),
	(rtsp_message_add_header), (rtsp_message_remove_header),
	(rtsp_message_get_header), (rtsp_message_set_body),
	(rtsp_message_take_body), (rtsp_message_get_body),
	(rtsp_message_steal_body), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Reorder arguments, object goes as the first one.
	Use g_return_val some more.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Export sometimes source pad with correct caps on the template, create
	the ghostpad from the template.
	Remove RTCP template as we never expose RTCP.
	Protect against invalid body size.
	Avoid memcpy when creating the output buffer.
	Properly post an error and send EOS when the loop function is shut down.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Make sure we can never set an invalid location.

	* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
	* gst/rtsp/rtspmessage.h:
	Added _steal_body method for future use.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
	Make freeing of NULL url return immediatly.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Use boilerplate.
	Make rtspsrc subclass GstBin to make state changes easier.
	Add Range header field on the PLAY request.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
	* gst/rtsp/rtspconnection.c: (inet_aton):
	Small cleanups.
	when multicast is selected as the transport, create UDP sources and
	connect to the multicast group.
	Move parsing and setting of caps to a common place.
	Fixes #349894.

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2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* ext/flac/gstflactag.c:
	* gst/alpha/gstalpha.c:
	* gst/debug/breakmydata.c:
	* gst/debug/negotiation.c:
	* gst/debug/testplugin.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videomixer/videomixer.c:
	* sys/sunaudio/gstsunaudiosrc.h:
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	More G_OBJECT macro fixing.
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2006-09-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Yves Lefebvre <ivanohe at abacom dot com>

	* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
	Correctly set the dwLength in strh.
	With this patch, the file duration is now displayed correctly in window
	media player and the AVI plays completely. Fixes #356147

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	Patch by: Darren Kenny <darren dot kenny at sun dot com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list):
	Set the output track as the MASTER so that the gnome-settings-daemon
	keybindings for changing the volume using the keyboard works.
	Fixes #356142.

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	Fix documentation, it is not possible to control the framerate of jpegdec
	using filtered caps yet. Fixes #355210.
	Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
	stop when there is an error.

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2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't interpret a first buffer with an offset of NONE as
	  'from the middle of the stream', but only a first buffer
	  that has a valid buffer offset that's non-zero (see #345449).

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2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
	(gst_icydemux_typefind_or_forward):
	* gst/icydemux/gsticydemux.h:
	  When we merge/collect multiple incoming buffers for typefinding
	  purposes, keep an initial 0 offset on the first outgoing buffer
	  as well (otherwise id3demux won't work right). Fixes #345449.
	  Also Make buffer metadata writable before setting buffer caps.

	* tests/check/elements/icydemux.c: (typefind_succeed),
	(cleanup_icydemux), (push_data), (GST_START_TEST),
	(icydemux_suite):
	  Small test case for the above.

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2006-09-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  More code reuse and better logging in _peek_chunk(). Reintroduce check
	  for chunk sizes before reading them (avoid oom). Better handling for 
	  invalid chunksizes when streaming.

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2006-09-11  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_set_property):
	* gst/level/gstlevel.h:
          Fix type mixup in level->interval (gdouble<->guint64). Spotted by
          René Stadler

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2006-09-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_data):
	  Revert one change to fix streaming avi (adapter size != data size).

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Frédéric Riss  <frederic.riss at gmail dot com>

	* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
	(gst_matroska_demux_reset),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.h:
	  Add support for VOBSUB subtitle tracks and zlib-compressed
	  tracks. Make sure we start on a keyframe after a seek. (#343348)

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
	(gst_matroska_demux_push_flac_codec_priv_data),
	(gst_matroska_demux_push_xiph_codec_priv_data),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	* gst/matroska/matroska-ids.h:
	  Add basic FLAC support (#311586), not perfect yet though, needs some
	  tweaking in flacdec; also, seeking could be better.
	  Do better bounds checking when deserialising vorbis stream headers
	  to make sure we don't read beyond the end of the buffer on bad input.

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2006-09-04  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
	Seeking back in a file containing a CMML stream errors out if the seek
	goes back up to the CMML headers. This is because after the seek the xml
	processing instruction <?xml ...?> is submitted to the xml parser again, 
	which results in an error. The attached patch fixes the problem. 
	Fixes #353908.

	* ext/annodex/gstcmmlenc.h:
	Fix authors name.


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	* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
	New helper function to lessen the ifdefs.
	(GST_INFO_OBJECT): 
	(gst_dv1394src_iso_receive): Use it.
	(gst_dv1394src_create): Also use the control sockets in iec61883
	mode.
	(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
	handle for AVC operations; fixes #348233.

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2006-08-27  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audiofxgood.xml:
	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c:
	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c:
	* gst/audiofxgood/audiopanorama.c:
	* gst/audiofxgood/audiopanorama.h:
          Rename again (audiofxgood -> audiofx).

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2006-08-27  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_stream_scan):
          Initialze variables.

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2006-08-25  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_init), (gst_avi_demux_finalize),
	(gst_avi_demux_reset), (gst_avi_demux_index_last),
	(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
	(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	(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_stream_header_pull), (gst_avi_demux_do_seek),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
	(gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	More attempts to turn this into readable code.
	Don't leak adapters.
	Calculate duration according to index more efficiently.
	Don't try to act like we drive the pipeline in chain mode.

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	* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt):
	Fix build.

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2006-08-25  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
	Do some extra sanity checks.
	Fixes #350340.

	* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
	(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
	(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
	Check if clip->start_time is valid before adding the clip to the
	track list.
	Reset enc->preamble going from PAUSED to READY.
	Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
	only used for EOS.
	Only post an error message if we were the one that created the fatal
	GstFlowReturn value.

	* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
	(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
	Parse the seconds field of the npt-sec time format using %llu rather than
	%d and check that the value scaled by GST_SECOND doesn't overflow.
	Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
	Lookup a clip's track with clip->track rather than clip->id which
	makes no sense.
	Identify a clip by its track and start time and not its xml id.
	do some more input checking and make sure we don't do undefined shifts.

	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
	(cmml_tag_message_pop), (check_headers), (push_clip_full),
	(push_clip), (push_empty_clip), (check_output_clip),
	(GST_START_TEST), (cmmldec_suite):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
	(check_headers), (push_clip), (check_clip_times), (check_clip),
	(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
	Added some more checks.

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2006-08-24  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_transform_m2s_int),
	(gst_audio_panorama_transform_s2s_int),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float):
	* gst/audiofxgood/audiopanorama.h:
	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
          Make also the pan-property float (saves scaling and yields better
          resolution)

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	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float):
          ChangeLog surgery to add cymax's real name


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

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	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_int),
	(gst_audio_panorama_transform_s2s_int),
	(gst_audio_panorama_transform_m2s_float),
	(gst_audio_panorama_transform_s2s_float),
	(gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
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	* gst/audiofxgood/audiopanorama.c:
	(gst_audio_panorama_transform_m2s):
	  Fix docs & debug category. Add Fixme for volume pan levels.

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	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	  unbreak AVI index handling, some more debug, remove an obsolete
	  adapter_flush that caused streaming to wander off in the wild

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2006-08-24  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (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_odml),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull):
	* gst/avi/gstavidemux.h:
	Some more cleanups. 
	Fix totalFrames parsing in ODML.
	Disable use of index for length calculation in case of ODML as this is
	broken now.

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	* ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata):
	  Use libgsttag helper function here too.

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2006-08-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_init), (gst_avi_demux_dispose),
	(gst_avi_demux_reset), (gst_avi_demux_index_next),
	(gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	(gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk),
	(gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull),
	(gst_avi_demux_parse_subindex),
	(gst_avi_demux_read_subindexes_push),
	(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
	(sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_peek_tag),
	(gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain),
	(gst_avi_demux_sink_activate), (gst_avi_demux_activate_push),
	(gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	  Initial streaming support for avidemux (fixes #336465)

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  There is no taglibmux element ...

	* gst/rtsp/gstrtspsrc.c:
	  Use '%' rather than '&perc;' in gtk-doc blurb, docs build
	  was complaining about unknown entity here.

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2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
	(gst_avi_demux_process_next_entry):
	* gst/avi/gstavidemux.h:
	Mark DISCONT.
	Remove old unused fields and reorder the struct a bit.

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2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	* sys/oss/gstosssink.c: (gst_oss_sink_open),
	(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
	Small documentation updates.

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2006-08-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
	(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
	(gst_avi_demux_next_data_buffer),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
	(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
	(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
	* gst/avi/gstavidemux.h:
	Precalc most of the duration query for each stream.
	Make seeking more correct.
	Use GstSegment to track position and duration.
	Code cleanups and leak fixes.
	Calculate correct total duration based on index length.

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	* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
	(parse_insert_string_field):
	  If strings in text fields are marked ISO8859-1, but contain
	  valid UTF-8 already, then handle them as UTF-8 and ignore
	  the encoding. (#351794)

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
	(gst_flac_dec_write), (gst_flac_dec_loop),
	(gst_flac_dec_sink_event), (gst_flac_dec_chain),
	(gst_flac_dec_src_query):
	* ext/flac/gstflacdec.h:
	  Make flac-in-ogg work (#352100).

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
	  Don't unref buffers of which we've already given away
	  ownership to the adapter.

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2006-08-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
	  Make metadata extraction actually work.

	* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
	(gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
	(gst_speexenc_chain):
	  Fix metadata writing: replace old code which wrote completely
	  broken tags with libgsttag-based code. Plus miscellaneous
	  code cleanups (use static pad templates etc.) and a bunch
	  of leak fixes.

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2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiopanorama/.cvsignore:
	* gst/audiopanorama/Makefile.am:
	* gst/audiopanorama/audiofx.c:
	* gst/audiopanorama/audiopanorama.c:
	* gst/audiopanorama/audiopanorama.h:
          die! die! die! you should never have been there

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2006-08-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
	Fix invalid memory access in audiopanorama test suite.

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2006-08-21  Edward Hervey  <edward@fluendo.com>

	* tests/check/elements/.cvsignore:
	ignore built file

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2006-08-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	Fix the build again.

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2006-08-21  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c: (plugin_init):
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_get_unit_size),
	(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s),
	(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
	  resubmit with the desired name *again*

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2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
	* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
          use g_assert in _get_unit_size

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2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-audiofxgood.xml:
          cleanup -unused.txt to make it useful, add previously missing docs

	* ext/Makefile.am:
	* ext/esd/esdmon.c:
	* ext/esd/esdsink.c:
	* ext/esd/gstesd.c: (plugin_init):
          reflow to get rid of two external symbols

	* gst/audiofxgood/audiofx.c: (plugin_init):
          re-add

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2006-08-20  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/audiofxgood/.cvsignore:
	* gst/audiofxgood/Makefile.am:
	* gst/audiofxgood/audiofx.c
	* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property),
	(gst_audio_panorama_get_unit_size),
	(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s),
	(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
	* gst/audiofxgood/audiopanorama.h:
	* tests/check/Makefile.am:
	* tests/check/elements/audiopanorama.c: (setup_panorama_m),
	(setup_panorama_s), (cleanup_panorama), (GST_START_TEST),
	(panorama_suite), (main):
        Add audiofxgood plugin with audiopanorama element

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2006-08-18  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/Makefile.am:
	More Oss docs fixage. 

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2006-08-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_base_init),
	(gst_rtp_sv3v_depay_class_init), (gst_rtp_sv3v_depay_init),
	(gst_rtp_sv3v_depay_finalize), (gst_rtp_sv3v_depay_setcaps),
	(gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_set_property),
	(gst_rtp_sv3v_depay_get_property),
	(gst_rtp_sv3v_depay_change_state),
	(gst_rtp_sv3v_depay_plugin_init):
	* gst/rtp/gstrtpsv3vdepay.h:
	Added experimental SVQ3 depayloader.

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2006-08-18  Edward Hervey  <edward@fluendo.com>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
	(gst_dvdemux_loop), (gst_dvdemux_change_state):
	* ext/dv/gstdvdemux.h:
	When handling seek requests, don't send the newsegment event from the
	calling thread. Instead save it so it can be sent from the streaming
	thread.

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2006-08-17  Wim Taymans  <wim@fluendo.com>

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

	* gst/multipart/multipartdemux.c: (multipart_parse_header):
	Accept leading whitespace before the boundary
	This patch makes the demuxer allow some whitespace before the actual
	boundary. This makes the demuxer work with the ``old'' gstreamer
	multipartmuxer again (which placed an extra \n before the start
	of the stream) Fixes #349068.

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2006-08-17  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
	Error out on non-implemented stuff.

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2006-08-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Andy Wingo <wingo at pobox dot com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setup),
	(gst_signal_processor_start), (gst_signal_processor_stop),
	(gst_signal_processor_cleanup), (gst_signal_processor_setcaps),
	(gst_signal_processor_pen_buffer), (gst_signal_processor_flush),
	(gst_signal_processor_do_pulls), (gst_signal_processor_do_pushes),
	(gst_signal_processor_change_state):
	Make ladspa elements reusable. Fixes #350006.

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2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
	Convert ' ' into '_'. Try to keep as many characters in the padtemplate
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	names as possible. Fixes #349901.
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2006-08-16  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
	(gst_signal_processor_do_pushes):
	A push() gives away our refcount so we should not use the buffer on the
	pen anymore.

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2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	(gst_oss_mixer_element_finalize):
	  Don't leak device string.

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2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Require CVS of GStreamer core and -base (for
	  GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).

	* ext/taglib/gstid3v2mux.cc:
	  Write extended comment tags properly (#348762).

	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
	(parse_comment_frame):
	  Extract COMM frames into extended comments, which makes it
	  easier to properly retain the description bit of the tag
	  and maintain this information when re-tagging (#348762).

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2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Don't try to run annodex unit tests if the annodex
	  plugin has not been built (Fixes #351116).

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2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_find_best):
	  When we can't find a usable audiosink, don't error out,
	  but use a fake sink instead and post a warning message
	  on the bus (#341278).

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2006-08-16  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init):
	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	  Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for
	  ossmixer's new device property.

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  Add docs for OSS elements.

	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update to CVS version.
	  
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	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	Caps extra properties must be defined as strings for
	depayloaders because they are generated from an SDP.

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
	(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
	(gst_rtp_h264_depay_finalize), (decode_base64),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	(gst_rtp_h264_depay_set_property),
	(gst_rtp_h264_depay_get_property),
	(gst_rtp_h264_depay_change_state),
	(gst_rtp_h264_depay_plugin_init):
	* gst/rtp/gstrtph264depay.h:
	Added basic, not completely functional RFC 3984 H264 depayloader.

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	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
	Add pads after setting them up.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_finalize),
	(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Fix interleaved mode.
	 - Protect streaming with lock.
	 - Combine flows
	 - set caps on outgoing buffers.
	 - strip trailing \0 from data packets.
	 - Configure RTP/RTCP in stream.
	Use DEBUG_OBJECT more.

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	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	Turn a g_print into a DEBUG line.

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2006-08-13  Wim Taymans  <wim@fluendo.com>

	* sys/oss/gstossmixer.c: (gst_ossmixer_open), (gst_ossmixer_new):
	* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
	(gst_oss_mixer_element_init), (gst_oss_mixer_element_set_property),
	(gst_oss_mixer_element_get_property),
	(gst_oss_mixer_element_change_state):
	* sys/oss/gstossmixerelement.h:
	Small cleanups. Better error reporting.
	Add device property for the mixer instead of the hardcoded
	/dev/mixer. Fixes #350785.
	API: GstOssMixerElement::device property

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2006-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jens Granseuer <jensgr at gmx net>

	* gconf/Makefile.am:
	  Make --disable-schemas work right (they still need
	  to be copied to the installation directory, just not
	  applied). Fixes #351347 (also #344100).
	  
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2006-08-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac: back to HEAD

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=== release 0.10.4 ===

2006-08-14  Thomas Vander Stichele <thomas at apestaart dot org>

	* configure.ac:
	  releasing 0.10.4, "Dear Leader"

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2006-08-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	Patch by: Edward Hervey <edward@fluendo.com>

	* configure.ac:
	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_data):
	Send the newsegment event in the streaming thread.
	Fixes #347529

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2006-08-08  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
	(gst_smokeenc_resync), (gst_smokeenc_chain):
	  Refuse sink caps in the encoder if width or height is not a
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	  multiple of 16, the encoder does not support that yet (#349939);
	  along the same lines, check the return value of the encoder
	  setup function; also remove some debug log clutter.
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	* ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
	whether a processor can work in place or not, and for keeping
	track of its state. Change the FlowReturn instance variable from
	"state" to "flow_state", all callers changed.

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
	(gst_signal_processor_start, gst_signal_processor_stop)
	(gst_signal_processor_cleanup): New functions to manage the
	processor's state.
	(gst_signal_processor_setcaps): start() as well as setup() here.
	(gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
	(gst_signal_processor_change_state): Stop and cleanup the
	processor as we go to NULL.

	* ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
	INPLACE_BROKEN is not set.

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
	Do the alloc_buffer in bytes, not frames.
	
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2006-08-04  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	Fix rgb masks when recording in < 24bpp.

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2006-08-04  Andy Wingo  <wingo@pobox.com>

	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
	(gst_signal_processor_prepare)
	(gst_signal_processor_update_inputs)
	(gst_signal_processor_process, gst_signal_processor_pen_buffer)
	(gst_signal_processor_flush)
	(gst_signal_processor_sink_activate_push)
	(gst_signal_processor_src_activate_pull)
	(gst_signal_processor_change_state): Remove the last of the code
	that assumes that we process whole buffers at a time. Fix some
	debugging. Seems to work now in some cases.
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	(gst_signal_processor_src_activate_pull): BPB
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	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
	Fix nframes-choosing.
	(gst_signal_processor_init): Init pending_in and pending_out.

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	* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
	more default sample rate, although we never check that the sample
	rate actually gets set. Something for the future.
	(gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
	(gst_signal_processor_event): Refcount fixen.
	(gst_signal_processor_process): Pull the number of frames to
	process from the sizes of the buffers in the input pens.
	(gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
	(gst_signal_processor_do_pulls): Add an nframes argument, and use
	it instead of buffer_frames.
	(gst_signal_processor_getrange): Refcount fixen, pass nframes on
	to do_pulls.
	(gst_signal_processor_chain)
	(gst_signal_processor_sink_activate_push)
	(gst_signal_processor_src_activate_pull):  Refcount fixen.

	* ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.

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	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	(gst_signal_processor_process):
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	  don't query buffer-frames from caps, add lots of debug-log,
	  try fix for assert (#349189)
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2006-07-31  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c:
	Fix docs.

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	* ext/ladspa/gstsignalprocessor.c:
	(gst_signal_processor_add_pad_from_template),
	(gst_signal_processor_init), (gst_signal_processor_setcaps),
	(gst_signal_processor_process), (gst_signal_processor_pen_buffer),
	(gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
	(gst_signal_processor_sink_activate_push),
	(gst_signal_processor_src_activate_pull),
	(gst_signal_processor_change_state):
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	 Add debugs logs here and there, add more error handling, add some
	 FIXME comments, filed #349189
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	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
	(gst_smokeenc_setcaps), (gst_smokeenc_chain):
	Set caps on buffer correctly.  Fixes bug #349155.

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2006-07-28  Wim Taymans  <wim@fluendo.com>

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

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
	(gst_multipart_demux_class_init), (gst_multipart_demux_init),
	(gst_multipart_demux_finalize), (get_line_end),
	(multipart_parse_header), (multipart_find_boundary),
	(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
	(gst_multipart_set_property), (gst_multipart_get_property):
	Uses GstAdapter instead of own buffering.
	Actually parses the mime-type correctly (In tests the mime-type was
	always "" with the old version).
	Uses the Content-length header if available to speed up things.
	Reliably autoscans the boundary name by default.
	Fixes #349068.

	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	Don't start the stream with a \n.

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2006-07-28  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	  Open source with O_NONBLOCK (#349015).

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	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
	(gst_avi_demux_massage_index):
	* gst/avi/gstavidemux.h:
	  Whitespace fixes and more debug

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	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_create_element_with_pretty_name),
	(gst_auto_audio_sink_find_best),
	(gst_auto_audio_sink_change_state):
	  Get rid of old and unused magic sound-server properties stuff.
	  Add suffix to child sink's name that makes it easy to see from
	  the name alone which type it actually is (alsa, oss, esd, etc.).

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	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_set_property), (gst_udpsrc_get_property),
	(gst_udpsrc_start):
	* gst/udp/gstudpsrc.h:
	Rename "buffer" to "buffer-size" to make clear it is a size we set and
	not some sort of feature we enable.

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	* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  Use CLOSE_SOCKET() here instead of close() to maintain
	  win32 workiness.

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	Patch by: Thijs Vermeir <thijs dot vermeir at barco 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_udpsrc_start):
	* gst/udp/gstudpsrc.h:
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	Update documentation.
	Small cleanups. Fixes #348752.
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	API: buffer-size property
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2006-07-26  Wim Taymans  <wim@fluendo.com>

	Patch by: Kai Vehmanen <kv2004 at eca dot cx>

	* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	(gst_rtp_pcma_pay_handle_buffer):
	* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush),
	(gst_rtp_pcmu_pay_handle_buffer):
	Fix timestamp calculation on outgoing RTP packets.
	Fixes #348675.

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2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Fix writing of comment frames (should be COMM not TCOM),
	  is still sub-optimal though, since we don't retain or
	  extract the comment descriptions properly (#334375,
	  also see #334375).

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2006-07-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c:
	  #define 'fact' RIFF chunk if we are not compiling against
	  -base CVS (we don't want to depend on -base CVS for this
	  one define only, and also not for release order reasons).

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	* ext/taglib/gstid3v2mux.cc:
	  Handle multiple tags of the same type properly. Re-inject
	  unparsed ID3v2 frames that we get as binary blobs from
	  id3demux into the tag again so we don't lose information
	  when retagging (#334375).

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2006-07-25  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init):
	  Document newly-added properties properly, so that there is a
	  'Since: 0.10.4' in the plugin docs. Convert some property
	  names into canonical GObject style (GObject will do that
	  internally anyway).

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	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist):
	  Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
	  well, and add the version to the blob's buffer caps, since that
	  information will be needed for deserialisation later on (#348644).

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	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
	(gst_avi_demux_parse_stream):
	 Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
	 indentation and spacing.

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2006-07-24  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/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update files to CVS/Prerelease version, add esdsink docs.

	* ext/esd/esdsink.c:
	  Add gtk-doc blurb.

	* gst/rtp/gstrtpmp4vpay.c:
	  Fix typo in element description.

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	* ext/esd/esdsink.c: (gst_esdsink_open),
	(gst_esdsink_factory_init):
	  Prevent libesd from auto-spawning a sound daemon if it
	  is not already running. Now that we don't do evil stuff
	  like that any longer we can give esdsink a rank so that
	  autoaudiosink will try it as well if all other audio
	  sinks fail (#343051).

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	* ext/esd/Makefile.am:
	  Oops, need to remove README from EXTRA_DIST as well.

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	* ext/esd/README:
	  Remove, it contains nothing useful anyway.

	* ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
	(gst_esdsink_delay):
	  Some small clean-ups; use GST_BOILERPLATE etc.

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	* gst/law/alaw-decode.c: (alawdec_getcaps):
	* gst/law/alaw-encode.c: (alawenc_getcaps), (gst_alawenc_chain):
	* gst/law/mulaw-decode.c: (mulawdec_getcaps):
	* gst/law/mulaw-encode.c: (mulawenc_getcaps):
	Fix negotiation to deal with ANY/EMPTY caps instead of leaking.

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	* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
	(gst_wavparse_other), (gst_wavparse_perform_seek),
	(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	(gst_wavparse_pad_query):
	* gst/wavparse/gstwavparse.h:
	  Use information from 'fact' chunk for length calculation of compressed
	  samples. Calculate bps if bogus value is found in wav header (embeded
	  mp2/mp3).
	  

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	Based on patch by: Joni Valtanen  <joni dot valtanen at movial fi>

	* configure.ac:
	* gst/udp/Makefile.am:
	* gst/udp/gstdynudpsink.c: (gst_dynudpsink_init),
	(gst_dynudpsink_finalize), (gst_dynudpsink_close):
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init),
	(gst_multiudpsink_finalize), (gst_multiudpsink_close):
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudp.c: (plugin_init):
	* gst/udp/gstudpsink.h:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
	(gst_udpsrc_start), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	* gst/udp/gstudpnetutils.c: (gst_udp_net_utils_win32_inet_aton),
	(gst_udp_net_utils_win32_wsa_startup):
	* gst/udp/gstudpnetutils.h:
	  Port udp plugin to win32 (#345288).

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	* gst/rtsp/rtspconnection.c: (rtsp_connection_send):
	Remove unwanted DEBUG line.

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	* gst/id3demux/gstid3demux.c: (plugin_init):
	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist):
	* gst/id3demux/id3tags.h:
	  On second thought, it might be wiser and more efficient
	  not to do tag registration from a streaming thread.

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2006-07-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c:
	(id3demux_add_id3v2_frame_blob_to_taglist),
	(id3demux_id3v2_frames_to_tag_list):
	  Put ID3v2 frames we can't parse as binary blobs into private
	  tags, so that they are not lost when retagging, at least once
	  id3v2mux has been taught to re-inject those frames again.
	  See bug #334375.

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	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_process_next_entry):
	Fix some leaks.

	* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
	Don't use \n in debug lines.

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	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
Thomas Vander Stichele's avatar
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	  Add annodex and icydemux, cleanup the sections a bit
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2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

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

	* ext/taglib/gstid3v2mux.cc:
	  Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
	  ID3v2 TSSE frames (#347898).

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	* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
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	  Respect mpegversion for "video/mpeg" and give message in case of
	  unhandled versions.
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	* ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip),
	(gst_pngdec_caps_create_and_set), (gst_pngdec_task),
	(gst_pngdec_chain), (gst_pngdec_sink_event),
	(gst_pngdec_libpng_init), (gst_pngdec_change_state),
	(gst_pngdec_sink_activate_push):
	* ext/libpng/gstpngdec.h:
	Use statically allocated segment instead of leaking.
	Various cleanups.
	Fix flush and seek handling.

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	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init),
	(gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init),
	(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process),
	(gst_rtp_mp4g_depay_set_property),
	(gst_rtp_mp4g_depay_get_property),
	(gst_rtp_mp4g_depay_change_state),
	(gst_rtp_mp4g_depay_plugin_init):
	* gst/rtp/gstrtpmp4gdepay.h:
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
	(gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps),
	(gst_rtp_mp4g_pay_flush):
	Added simple generic mpeg4 depayloader.
	Fix generic mpeg4 payloader.

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2006-07-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
	  Don't try doing state changes on a NULL pointer.

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	Patch by: Sebastien Cote <sebas642 at yahoo dot ca>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_base_init),
	(gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_init),
	(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpamrdepay.h:
	rtpamrdec isn't a subclass of GstBaseRtpDepayload.
	Fixes #321191

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2006-07-14  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_get_caps), (gst_ximage_src_class_init):
	Fix segfault when moving mouse pointer to the bottom right corner.

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2006-07-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init),
	(gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init),
	(gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process),
	(gst_rtp_mp2t_depay_set_property),
	(gst_rtp_mp2t_depay_get_property),
	(gst_rtp_mp2t_depay_change_state),
	(gst_rtp_mp2t_depay_plugin_init):
	* gst/rtp/gstrtpmp2tdepay.h:
	Added mpeg2 TS depayloader. Closing #347234.

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2006-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_close):
2069
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2071
	  Remove g_assert that shouldn't be there and was triggered
	  after trying to open a device that doesn't exist or can't
	  be opened for some other reason (#347972).
2072

2073
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2080
2006-07-10  Edward Hervey  <edward@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_stream_header), (push_tag_lists):
	* gst/avi/gstavidemux.h:
	Don't push tag events found by gst_riff_parse_info() before outputting
	GST_EVENT_NEWSEGMENT.

2081
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2089
2006-07-10  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
	(rtsp_connection_close):
	* gst/rtsp/rtspdefs.h:
	replaced closesocket and close in code with one CLOSE_SOCKET. 
	Some more cleanups. Fixes #345301.

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2006-07-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	  Fix example pipeline in docs.

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2006-07-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Rob Taylor <robtaylor at floopily dot org>

	* gst/udp/gstmultiudpsink.c: (join_multicast),
	(gst_multiudpsink_init_send), (gst_multiudpsink_add):
	If a destination is added before the stream is set to PAUSED, the
	multicast group is not joined as the socket is not created yet. 
	Also TTL and LOOP should also be set. Fixes #346921.

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2006-07-09  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_set_property), (gst_ximage_src_get_property),
	(gst_ximage_src_get_caps), (gst_ximage_src_class_init),
	(gst_ximage_src_init):
	* sys/ximage/gstximagesrc.h:
	Fix use-damage property to actually work :)
	Add startx, starty, endx, endy properties so screencasts other than full
	screen ones can work.

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2006-07-08  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
	(gst_ximage_src_set_property), (gst_ximage_src_get_property),
	(gst_ximage_src_class_init), (gst_ximage_src_init):
	* sys/ximage/gstximagesrc.h:
	Add use_damage property to offer ability to choose whether to use
	XDamage or not.

2125
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2130
2006-07-07  Wim Taymans  <wim@fluendo.com>

	* gst/goom/filters.c: (zoomFilterSetResolution):
	Avoid goom coredumping by clearing memory. 
	Fixes 345679.

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2006-07-05  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/libgstid3demux.dsp:
	Add a link to libgsttag-0.10.lib.

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2006-07-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	(gst_tag_demux_read_range):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	(gst_id3demux_read_range):
	  Don't return FLOW_UNEXPECTED when a buffer is before
	  the start of the stream (which might happen with
	  large ID3v2 tags if the tag reading was done pullrange
	  based and we then switched to push mode later on).
	  Fixes regression introduced by commit from June 29th.

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2006-07-05  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Make UTF-8 the default encoding when writing string
	  tags (before, our UTF-8 strings would automatically
	  be converted to ISO-8859-1 by taglib and written as
	  ISO-8859-1 fields if that was possible).

	* tests/check/elements/id3v2mux.c: (utf8_string_in_buf),
	(test_taglib_id3mux_check_tag_buffer), (identity_cb),
	(test_taglib_id3mux_with_tags):
	  Add test case that makes sure our UTF-8 strings have
	  actually been written into the tag as UTF-8.

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2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Let's try that again.

2167
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2006-07-04  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Disable monoscope plugin for now until it fulfills
	  all the requirements.

2173
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2006-07-03  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	* gst/monoscope/Makefile.am:
	* gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init),
	(gst_monoscope_class_init), (gst_monoscope_init),
	(gst_monoscope_finalize), (gst_monoscope_reset),
	(gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps),
	(gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain),
	(gst_monoscope_sink_event), (gst_monoscope_src_event),
	(gst_monoscope_change_state), (plugin_init):
	* gst/monoscope/gstmonoscope.h:
	  Port monoscope visualisation to 0.10.

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2006-07-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Return FLOW_UNEXPECTED when at the end of the file, not
	  FLOW_ERROR. Fixes 'internal stream error' errors that
	  would sometimes occur in totem when scrubbing to the
	  end of an ID3v1 tagged mp3 file.

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2006-07-03  Edward Hervey  <edward@fluendo.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback),
	(buffer_clip), (user_end_callback), (gst_pngdec_chain),
	(gst_pngdec_sink_event), (gst_pngdec_change_state):
	* ext/libpng/gstpngdec.h:
	Implement buffer clipping/dropping using GstSegment.
	This provides accurate seeking.

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2006-07-03  Edward Hervey  <edward@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
	(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
	(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
	(gst_avi_demux_process_next_entry), (push_tag_lists),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	* gst/avi/gstavidemux.h:
	Proper aggregation of each stream's GstFlowReturn in order to figure out
	whether the task should stop or not.
	Don't send inline events before pushing out a NEW_SEGMENT, more
	specifically for GST_TAG_EVENT.
	Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
	sub-indexes.

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2006-06-30  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list):
	  Move "Monitor" slider to input tab so it works more like
	  sdtaudiocontrol, which is what people on Solaris are used
	  to using for their mixer program (#346259).

2230
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2006-06-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/elements/level.c: (GST_START_TEST):
	  fix a leak, clean up at the end

2235
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2006-06-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	(gst_matroska_demux_send_event),
	(gst_matroska_demux_loop_stream_parse_id):
	* gst/matroska/matroska-ids.h:
	  Send tag event after newsegment event.

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2006-06-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
	(gst_id3demux_read_range):
	  Make sure we don't return GST_FLOW_OK with a NULL buffer in
	  certain cases where a read beyond the end of the file is
	  requested. Fixes #345930.

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
	(gst_tag_demux_read_range):
	  Fix same issue here as well.

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2006-06-29  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
	
	Fix hypothetical crash.

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2006-06-28  Tim-Philipp Müller  <tim at centricular dot net>

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

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Do not modify the ports value. If the user has turned off the
	  built-in speakers, then we should not reset it in the prepare
	  function, since this causes the built-in speakers to turn
	  back on anytime the user changes a track in totem, rhythmbox,
	  etc. (#346066).

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2006-06-23  Wim Taymans  <wim@fluendo.com>

	* gst/goom/gstgoom.c: (gst_goom_src_negotiate):
	Fix double caps unref when negotiation fails.

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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/annodex/gstcmmldec.c:
	* ext/annodex/gstcmmlenc.c:
	* ext/annodex/gstcmmlparser.c:
	* ext/dv/gstdvdec.c:
	* ext/dv/gstdvdemux.c:
	* ext/gdk_pixbuf/pixbufscale.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/gstsmokedec.c:
	* ext/jpeg/gstsmokeenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngenc.c:
	* ext/speex/gstspeexenc.c:
	* gst/alpha/gstalphacolor.c:
	* gst/cutter/gstcutter.c:
	* gst/debug/gstnavigationtest.c:
	* gst/icydemux/gsticydemux.c:
	* gst/level/gstlevel.c:
	* gst/multipart/multipart.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstvideoflip.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
	  plus two minor macro fixes.

2312
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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c:
	(gst_matroska_demux_check_subtitle_buffer),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.c:
	(gst_matroska_track_init_subtitle_context):
	* gst/matroska/matroska-ids.h:
	  Try to fix up broken matroska files containing subtitle
	  streams with non-UTF8 character encodings (courtesy of
	  mkvmerge) using either the encoding specified in the
	  GST_SUBTITLE_ENCODING environment variable or the
	  current locale's character set if it is non-UTF8.
	  Fixes #337076.

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2006-06-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Set image type from APIC frame as "image-type" field
	  of GST_TAG_IMAGE buffer caps (#344605).

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2006-06-20  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/Makefile.am:
	* ext/flac/gstflacdec.c: (gst_flac_dec_init),
	(gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize),
	(gst_flac_dec_metadata_callback),
	(gst_flac_dec_metadata_callback_seekable),
	(gst_flac_dec_metadata_callback_stream),
	(gst_flac_dec_error_callback),
	(gst_flac_dec_error_callback_seekable),
	(gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable),
	(gst_flac_dec_read_stream), (gst_flac_dec_write),
	(gst_flac_dec_write_seekable), (gst_flac_dec_write_stream),
	(gst_flac_dec_loop), (gst_flac_dec_sink_event),
	(gst_flac_dec_chain), (gst_flac_dec_convert_sink),
	(gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query),
	(gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
	(gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate),
	(gst_flac_dec_sink_activate_push),
	(gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
	* ext/flac/gstflacdec.h:
	  Support chain-based operation, should make flac-over-DAAP
	  work (#340492).

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2006-06-20  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	Doc updates, merge some unused symbols.

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2006-06-20  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	Added documentation for the rtsp plugin. Fixes #345393.

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2006-06-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
	(rtsp_connection_close), (rtsp_connection_free):
	Use better G_OS_* macros. Fixes #345301 some more.

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2006-06-20  Wim Taymans  <wim@fluendo.com>

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

	* sys/sunaudio/Makefile.am:
	* sys/sunaudio/gstsunaudio.c: (plugin_init):
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
	(gst_sunaudiomixer_ctrl_list_tracks),
	(gst_sunaudiomixer_ctrl_get_volume),
	(gst_sunaudiomixer_ctrl_set_volume),
	(gst_sunaudiomixer_ctrl_set_mute),
	(gst_sunaudiomixer_ctrl_set_record):
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.c:
	(gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
	(gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
	(gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
	(gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
	(gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
	(gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
	(gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
	(gst_sunaudiosrc_reset):
	* sys/sunaudio/gstsunaudiosrc.h:
	Add a SunAudio source plugin.
	Support stereo and right/left channel gain in the mixer plugin.
	Support the RECORD flag so that you can switch between line-input and
	microphone in gnome-volume-control.
	Code cleanups like using an enumerator for track number instead of an 
	integer. Fixes #344923.

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2006-06-20  Wim Taymans  <wim@fluendo.com>

	Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>

	* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
	(rtsp_connection_close):
	Make RTSP plugin compile on windows. Fixes #345301.
	Some changes to original patch to catch errors better.
	use ifdef WIN32 instead of ifndef.

2424
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2006-06-19  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* configure.ac:
	If we have libraw1394 >= 1.2.1, then we need libiec61883.

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2006-06-18  Edward Hervey  <edward@fluendo.com>

	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): 
	After a failed buffer alloc, we need to abort the jpeg decoding (it
	started when parsing headers to figure out how many bytes we need
	to request downstream).

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2006-06-18  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
	  Make sure we don't read beyond the end of the file (#345232).

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2006-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix --disable-external (can't set conditionals conditionally,
	  #343602).

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