ChangeLog 493 KB
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2007-08-20  Wim Taymans  <wim.taymans@gmail.com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* MAINTAINERS:
	Updating all the maintainers files

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2007-06-22  Edward Hervey  <edward@fluendo.com>

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

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

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

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

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

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

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

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

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

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

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

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

	* configure.ac:
	Back to CVS

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

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=== release 0.10.6 ===

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

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

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

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

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

	Patch by: Peter Kjellerstedt <pkj at axis com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* po/POTFILES.in:
	Add qtdemux for translation

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

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

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

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

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

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	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-directdraw.xml:
	* docs/plugins/inspect/plugin-directsound.xml:
	* docs/plugins/inspect/plugin-waveform.xml:
	Move the waveform plugin from -bad too. Update the inspect xml
	files to mention Plugins Good instead of Plugins Bad.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
	(gst_wavenc_chain), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Properly write wav files with width!=depth by having the depth most
	significant bytes set and all others zero. Fixes #442535.

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	* gst/rtsp/rtspconnection.c:
	Add include to make buildbot happy.

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	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (add_date_header),
	(rtsp_connection_send), (parse_response_status),
	(parse_request_line), (parse_line), (rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspmessage.c: (key_value_foreach),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_remove_header), (rtsp_message_append_headers),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Improves version checking, allowing an RTSP server to reply with "505
	RTSP Version not supported.
	Adds a Date header to all messages.
	Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
	want to be able to send a response even if something in the request was
	invalid. EINVAL is only used when passing wrong arguments to functions.
	Do not handle an invalid method in parse_request_line(). Defer this to
	the caller so it can respond with "405 Method Not Allowed".
	Improves parsing of the timeout parameter to the Session header,
	allowing whitespace after the semicolon. 
	Avoids a compiler warning due to variables shadowing a function argument.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-05-14  Wim Taymans  <wim@fluendo.com>

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

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

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2007-05-14  Wim Taymans  <wim@fluendo.com>

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

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2007-05-14  Wim Taymans  <wim@fluendo.com>

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

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2007-05-14  Wim Taymans  <wim@fluendo.com>

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

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

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2007-05-14  Wim Taymans  <wim@fluendo.com>

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

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

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

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

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

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

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

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

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

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

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

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

	Patch by: Eric Anholt

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

1685
1686
1687
1688
1689
1690
1691
2007-05-08  Tim-Philipp Müller  <tim at centricular dot net>

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

1692
1693
1694
1695
1696
1697
1698
1699
2007-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

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

1700
1701
1702
1703
1704
1705
1706
1707
1708
1709
1710
1711
1712
2007-05-04  Wim Taymans  <wim@fluendo.com>

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

1713
1714
1715
1716
1717
2007-05-04  Wim Taymans  <wim@fluendo.com>

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

1718
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1720
1721
1722
1723
1724
1725
1726
1727
1728
1729
1730
1731
1732
1733
1734
1735
1736
2007-05-04  Wim Taymans  <wim@fluendo.com>

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

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

1737
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1739
1740
1741
1742
2007-05-03  Wim Taymans  <wim@fluendo.com>

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

1743
1744
1745
1746
1747
1748
1749
2007-05-03  Wim Taymans  <wim@fluendo.com>

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

1750
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1753
1754
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1761
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1764
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1767
1768
1769
1770
2007-05-03  Wim Taymans  <wim@fluendo.com>

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

1771
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1776
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1778
1779
1780
1781
1782
1783
1784
1785
1786
1787
1788
1789
2007-05-02  Wim Taymans  <wim@fluendo.com>

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

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

1790
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1793
1794
1795
1796
2007-05-02  Wim Taymans  <wim@fluendo.com>

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

1797
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1800
1801
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

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

1802
1803
1804
1805
1806
1807
1808
1809
1810
1811
1812
2007-05-02  Wim Taymans  <wim@fluendo.com>

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

1813
1814
1815
1816
1817
1818
1819
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

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

1820
1821
1822
1823
1824
1825
2007-05-02  Wim Taymans  <wim@fluendo.com>

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

1826
1827
1828
1829
1830
1831
1832
1833
1834
1835
1836
1837
1838
1839
1840
1841
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1845
1846
1847
1848
1849
1850
1851
1852
1853
1854
1855
1856
1857
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.

1858
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1860
1861
1862
1863
1864
1865
1866
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.

1867
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1869
1870
1871
1872
1873
1874
1875
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.

1876
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1878
1879
1880
1881
1882
1883
1884
1885
1886
1887
1888
1889
1890
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.

1891
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1894
1895
1896
1897
1898
1899
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.

1900
1901
1902
1903
1904
1905
1906
1907
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.

1908
1909
1910
1911
1912
2007-04-27  Wim Taymans  <wim@fluendo.com>

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

1913
1914
1915
1916
1917
1918
1919
1920
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.

1921
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1923
1924
1925
1926
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1930
1931
1932
1933
1934
1935
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.

1936
1937
1938
1939
1940
1941
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.

1942
1943
1944
1945
1946
1947
1948
1949
1950
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.

1951
1952
1953
1954
1955
1956
1957
1958
1959
1960
1961
1962
1963
1964
1965
1966
1967
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.

1968
1969
1970
1971
1972
1973
1974
1975
1976
1977
1978
1979
1980
1981
1982
1983
1984
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.

1985
1986
1987
1988
1989
1990
1991
1992
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).

1993
1994
1995
1996
1997
2007-04-25  Wim Taymans  <wim@fluendo.com>

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

1998
1999
2000
2001
2002
2003
2004
2005
2006
2007
2008
2009
2010
2011
2012
2013
2014
2015
2016
2017
2018
2019
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.

2020
2021
2022
2023
2024
2025
2026
2027
2028
2029
2030
2031
2032
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.

2033
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2035
2036
2037
2007-04-25  Stefan Kost  <ensonic@users.sf.net>

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

2038
2039
2040
2041
2042
2043
2044
2045
2046
2047
2048
2049
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.

2050
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2052
2053
2054
2055
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).

2056
2057
2058
2059
2060
2061
2062
2063
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.

2064
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2066
2067
2068
2069
2070
2071
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.

2072
2073
2074
2075
2076
2077
2078
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.

2079
2080
2081
2082
2083
2084
2085
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.

2086
2087
2088
2089
2090
2091
2092
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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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	* 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.

Wim Taymans's avatar
Wim Taymans committed
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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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