ChangeLog 428 KB
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2007-05-02  Wim Taymans  <wim@fluendo.com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  Add minimal docs blurb to alphacolor; split out headers into
	  separate header file for gtk-doc.

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

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

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

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

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

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

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

	Patch by: Snaik <snaik32 gmail com>

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

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

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

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

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2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-04-12  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	Ported mulaw and alaw payloaders to use new base class

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Trigger rebuild.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* po/POTFILES.in:
	  Update.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-03-01  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

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

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

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

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

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

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

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

	* configure.ac:
	  Convert to new AG_GST style.

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

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

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

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

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

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

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

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

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

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

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

	Patch by: Mark Nauwelaerts <manauw at skynet be>

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

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

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

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

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

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2007-02-28  Michael Smith  <msmith@fluendo.com>

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

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2007-02-27  Christian Schallerr <christian@fluendo.com>

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

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2007-02-26  Jan Schmidt  <thaytan@mad.scientist.com>

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

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2007-02-26  Jan Schmidt  <thaytan@mad.scientist.com>

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

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

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

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

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

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

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

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

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

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2007-02-22  Edgard Lima <edgard.lima@indt.org.br>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
	(gst_gconf_render_bin_from_key),
	(gst_gconf_get_default_audio_sink):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
	(do_toggle_element), (gst_gconf_audio_sink_set_property),
	(gst_gconf_audio_sink_get_property):
	  In gconfaudiosink, get the right key as the old key in do_toggle
	  (ie. one dependent on the profile selected). Log some more stuff so
	  we can see what's actually going on.

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2007-02-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
	(gst_audio_amplify_class_init), (gst_audio_amplify_init),
	(gst_audio_amplify_set_process_function),
	(gst_audio_amplify_setup):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_setup):
	* gst/audiofx/audioinvert.h:
	Some small cleanups and port both elements to the new GstAudioFilter
	base class to save a few lines of common code.
	* gst/audiofx/Makefile.am:
	Link against libgstaudio for the above changes

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2007-01-29  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/.cvsignore:
	Some more ignores.

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2007-01-26  Wim Taymans  <wim@fluendo.com>

	Patch by: charles <charlesg3 at gmail dot com>

	* ext/shout2/gstshout2.c: (gst_shout2send_init),
	(set_shout_metadata), (gst_shout2send_event):
	* ext/shout2/gstshout2.h:
	Properly handle tags in shout2send. Fixes #399825.

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_activate_streams):
	Convert SDP fields to upper/lowercase following the rules in the SDP to
	caps document. 

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix case of encoding-name and key/value pairs to match the document.
	This is to make interoperation with SDP case-insensitive as required by
	the relevant RFCs.

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	Bump required -core/-base to CVS

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
	(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
	* gst/rtp/gstrtpL16pay.h:
	Fill up to MTU using adapter.
	Timestamp rtp packets.

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2007-01-25  Edward Hervey  <edward@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
	Use G_GSIZE_FORMAT in print statements for portability.
	Fixes build on macosx.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
	(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
	(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
	(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
	(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
	(gst_rtp_L16_depay_plugin_init):
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
	(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
	(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
	(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
	(gst_rtp_L16_pay_plugin_init):
	* gst/rtp/gstrtpL16pay.h:
	Port and enable raw audio payloader/depayloader. Needs a bit more work
	on the payloader side.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
	* gst/rtsp/gstrtspsrc.h:
	Only unblock the udp pads when we linked and activated them all.
	Fixes #395688.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
	(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
	(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
	(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
	(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
	* gst/rtp/gstrtpac3depay.h:
	Added simple AC3 depayloader (RFC 4184).

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	Fix a leak.

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2007-01-24  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audioamplify.c:
	(gst_audio_amplify_clipping_method_get_type),
	(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
	(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
	(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
	(gst_audio_amplify_set_caps),
	(gst_audio_amplify_transform_int_clip),
	(gst_audio_amplify_transform_int_wrap_negative),
	(gst_audio_amplify_transform_int_wrap_positive),
	(gst_audio_amplify_transform_float_clip),
	(gst_audio_amplify_transform_float_wrap_negative),
	(gst_audio_amplify_transform_float_wrap_positive),
	(gst_audio_amplify_transform_ip):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new element "audioamplify". This allows scaling of raw audio
	samples, similar to the "volume" element, but provides different modes
	for clipping and allows unlimited amplification. It's mainly targeted
	for creative sound design and not as a replacement of the "volume"
	element. Fixes #397162
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for audioamplify and integrate them into the build system
	* tests/check/Makefile.am:
	* tests/check/elements/audioamplify.c: (setup_amplify),
	(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
	Add fairly extensive unit test suite for audioamplify

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
	Unblock pads after adding the pads to the element so that autopluggers
	get a change to link something. Possibly fixes #395688.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
	(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
	(gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
	(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix caps with payload numbers.
	Add some fixed payload numbers to caps when possible.

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1987
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1989
2007-01-23  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c: (plugin_init):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_init),
	(gst_audio_invert_set_property), (gst_audio_invert_get_property),
	(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
	(gst_audio_invert_transform_float),
	(gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Add new audiofx element "audioinvert". This element swaps the upper
	and lower half of samples and can be used for example for a
	wide-stereo effect. Fixes #396057
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for the audioinvert element and add them to the build system.
	* tests/check/Makefile.am:
	* tests/check/elements/audioinvert.c: (setup_invert),
	(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
	Add unit test suite for the audioinvert element.

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2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
	(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
	Parse config params as string and int.
	Parse and use AU header length

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2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
	(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
	* gst/smpte/gstmask.c: (_gst_mask_register):
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
	* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
	(gst_smpte_paint_triangle_clock):
	constify some static structs.
	Don't update the mask if nothing changed to the params.
	Make sure we never draw outside of the picture. Fixes #398325.

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2013
2014
2015
2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
	  Error out properly when pull_range fails while we're reading the
	  headers, instead of just pausing the task silently. Fixes #399338.

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2023
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	  Some more sanity checks to make sure the input formats match and the
	  input pads are actually negotiated, in case someone tries to feed
	  buffers from fakesrc or filesrc. Fixes #398299.
	  Also const-ify an array, just because we can.

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2007-01-19  Edward Hervey  <edward@fluendo.com>

	* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
	Ignore previous commit, that was only valid for widths and heights
	that are multiples of 4.
	Copy over size/stride macros from jpegdec. This allows the element
	to work with any width,height...
	... but puts in evidence that the actual transformations only work
	with width/height that are multiples of 4.

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	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	Allocate buffers of the right size.
	The proper size of a I420 buffer in bytes is:
	
	    width * height * 3
	    ------------------
	            2

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2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_init):
	  Proxy getcaps on sink pads too, so that we either end up with the
	  same dimensions on all pads or error out if that's not possible
	  (seems to work even!). Fixes #398086, I think.

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2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
	  fix integer properties with -1 as minimum value.

	* 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-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-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.

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	Patch by: Sebastian Dröge <slomo circular-chaos org>

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	* gst/audiofx/audiopanorama.c:
	  Fix doc section name (Fixes #397946)

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2007-01-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2object.c:
	(gst_v4l2_object_install_properties_helper),
	(gst_v4l2_object_set_property_helper),
	(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_init), (gst_v4l2src_set_property),
	(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
	(gst_v4l2src_capture_deinit):
	  Fix EIO handing when capturing. Add new property to specify the number of
	  buffers to enque (and remove the borked num-buffers usage).

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2007-01-16  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
	(gst_audio_panorama_set_process_function):
	  Use a function array for process methods, add more docs and define the
	  startindex of enums.

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	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
	(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
	(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
	(gst_avi_mux_riff_get_avi_header),
	(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
	(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
	(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
	(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
	(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
	(gst_avi_mux_change_state):
	* gst/avi/gstavimux.h:
	* tests/check/elements/avimux.c: (teardown_src_pad):
	  Add support for more than one audio stream; write better AVIX
	  header; refactor code a bit; don't announce vorbis caps on our audio
	  sink pads since we don't support it anyway. Closes #379298.

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2007-01-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c:
	(gst_audio_panorama_method_get_type),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_process_function),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_int_simple),
	(gst_audio_panorama_transform_s2s_int_simple),
	(gst_audio_panorama_transform_m2s_float_simple),
	(gst_audio_panorama_transform_s2s_float_simple):
	* gst/audiofx/audiopanorama.h:
	  Add 'method' property and provide a simple (non-psychoacustic)
	  processing method (#394859).

	* tests/check/elements/audiopanorama.c: (GST_START_TEST),
	(panorama_suite):
	  Tests for new method.

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2007-01-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
	  Set correct caps on outgoing pulled buffers, or things blow up
	  after recent core changes.

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2007-01-11  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_init),
	(gst_multipart_mux_request_new_pad),
	(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Return FLOW errors ASAP. Fixes #394977.
	Misc cleanups.

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2007-01-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	Check for stream pad before activating. 

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	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/COPYING.MIT:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
	(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_open), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_send), (read_line),
	(parse_request_line), (parse_line), (rtsp_connection_read),
	(rtsp_connection_close):
	* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
	(rtsp_method_as_text), (rtsp_header_as_text),
	(rtsp_status_as_text), (rtsp_find_header_field),
	(rtsp_find_method):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
	(rtsp_ext_wms_configure_stream):
	* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
	(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_get_header), (rtsp_message_set_body),
	(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
	(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
	(sdp_message_dump):
	Allow url to be NULL to be able to use it for server connections.
	Can now send responses as well as requests.
	No longer hangs in an endless loop if EOF is received.
	Can now convert a status code to a text string.
	Return RTSP_HDR_INVALID for unknown headers.
	Return RTSP_INVALID for unknown methods.
	Copy CSeq and Session headers from the request.
	Only free memory corresponding to the currently set message type.
	Added const to function arguments as appropriate.
	Avoid a compiler warning when initializing nmedia.
	Use guint rather than gint to avoid compiler warnings.
	Fix crasher in wms extension.
	Factor out stream setup from open_connection.
	Delay activation of streams when actual data is received from the
	server, this prepares us to do proper protocol switching.
	Added new license.
	Fixes #380895.


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2007-01-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo ubuntu com>

	* docs/plugins/Makefile.am:
	* gst/audiofx/audiopanorama.c:
	  Some small docs fixes (#394851).

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2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c:
	Fix docs.

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2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
	(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
	(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
	(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
	(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
	* gst/rtp/gstrtpmpvdepay.h:
	  Added RFC 2250 MPEG Video Depayloader.

	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	Fix Header file. Small cleanups.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
	(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
	(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
	(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
	(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
	(gst_rtp_mp4v_depay_change_state):
	Remove usused code. Remove Adapter from state Change. Added debug.

	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
	(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
	(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpadepay.h:
	Subclass base depayloader.
	Added debug.
	Support static payload type assignment as well.

	* gst/rtp/gstrtpmpapay.c:
	Fix caps.

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2007-01-08  Tim-Philipp Müller  <tim at centricular dot net>

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

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/smokecodec.c:
	  These libjpeg callbacks should return a 'boolean' (unsigned char
	  apparently) and not a 'gboolean' (which maps to gint). Fixes
	  warnings when compiling with MingW (#393427).

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  Use ioctlsocket on win32.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Some printf format fixes for win32.

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2007-01-07  Sébastien Moutte  <sebastien@moutte.net>

	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	  Use gst_guint64_to_gdouble for conversion.
	* win32/vs6/libgstmatroska.dsp:
	  Add zlib to the link.
	* win32/vs6/libgstvideobox.dsp:
	  Update liboil library name (project is linked to 
	  liboil-0.3-0.lib now).
	  
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2007-01-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/Makefile.am:
	  If zlib is available and used, we must link it explicitly for
	  things to work on MingW (fixes #392855).

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2007-01-04  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/esdsink.c: (gst_esdsink_delay):
	  Don't return bogus values when esd_get_delay() fails for some
	  reason (#392189).

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2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/gstximagesrc.c: (composite_pixel):
	  Fix presumably copy'n'pasto for 16bpp depth.

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2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-mux.c:
	(gst_matroska_mux_audio_pad_setcaps):
	  The "signed" field in audio caps is of boolean type, trying to use
	  gst_structure_get_int() to extract it will fail. Fixing this makes
	  matroskamux accept raw audio input (#387121) (use at your own risk
	  though, due to the matroska spec being not entirely useful in this
	  respect).
	  Also fix up raw audio structures in template caps so that they
	  represent what our setcaps function will actually accept, so that
	  converters know what to convert to.
	  Finally, don't fail if there isn't an "endianness" field in 8-bit
	  PCM caps.

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2006-12-22  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  reapply consistent pad (de)activation

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2006-12-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.5 doap entry

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=== release 0.10.5 ===

2006-12-21  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.5, "The Path of Thorns"

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2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  revert my freeze breakage

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2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  consistent pad (de)activation

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2006-12-18  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* ext/Makefile.am:
	Disable LADPSA, as it has moved to the -bad module for the duration.

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2006-12-18  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	(gst_signal_processor_event):
	Reset flow_state back to _OK after a flush stop so that we exit our
	error state after the flush. Fixes #374213

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2006-12-16  David Schleef  <ds@schleef.org>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Decent effort at porting to 0.10.  Needs cleanup on OS/X.

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2006-12-16  David Schleef  <ds@schleef.org>

	Patch by: Vijay Santhanam <vijay santhanam gmail com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Preliminary patch for porting osxvideosink

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2006-12-16  Wim Taymans  <wim@fluendo.com>

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

	* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
	(gst_videomixer_set_master_geometry),
	(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
	(gst_videomixer_reset), (gst_videomixer_init),
	(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
	(gst_videomixer_release_pad), (gst_videomixer_collected),
	(gst_videomixer_change_state):
	Introduce some locking around the videomixer state so that it does not
	crash when adding/removing pads. Fixes #383043.

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2006-12-16  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Make sure libcaca can actually be used instead of just checking for
	  /usr/bin/caca-config, so we don't wrongly try to build cacasink when
	  cross-compiling (fixes #384587).

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
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2006-12-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	* gst-plugins-good.doap:
	* gst-plugins-good.spec.in:
	  adding doap file

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2006-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  libflac-1.1.3 changed API again, but we can't build against it yet,
	  so make sure our check doesn't use libflac-1.1.3 and add a comment
	  to this effect.

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