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

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

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

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

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

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

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

Stefan Kost's avatar
Stefan Kost committed
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2007-07-20  Stefan Kost  <ensonic@users.sf.net>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_caps_to_pixfmt):
	Fix the r_mask test for RGBA32 on little-endian.
	Fix a stupid typo that would have obviously broken 
	compilation on big-endian, if anyone was testing.

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2007-07-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
	(paint_hline_str4):
	* gst/videotestsrc/videotestsrc.h:
	Add alpha to the color struct.
	Use a default alpha value of 255 instead of 128.

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2007-07-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstplaybasebin.c: (no_more_pads_full),
	(setup_source):
	Clear the dynamic pads counter when starting a new uri. This makes
	reusing playbin work again.
	Fixes #454264.

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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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2007-07-12  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	* tests/check/elements/volume.c: (GST_START_TEST):
	  Fix 'make check' build against core CVS.

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2007-07-10  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/interfaces/propertyprobe.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Make gtk-doc happy.

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2007-07-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_callback):
	  Quick hack to make audiosinks stop at EOS when operating in
	  pull-mode; needs to be fixed properly some day.

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2007-07-06  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Fix location of includes in the docs.

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

	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	(gst_ffmpegcsp_avpicture_fill):
	* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
	(img_get_alpha_info):
	Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
	of the existing BGRA32 and RGBA32 formats with the alpha at the other
	end of the word. Partially fixes #451908

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2007-07-05  Stefan Kost  <ensonic@users.sf.net>

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

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2007-07-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
	(gst_adder_request_new_pad):
	Make getcaps more robust by not using the proxycaps function. This makes
	sure that we don't end up recursively calling getcaps upstream.
	See #316248.

2007-06-29  Wim Taymans  <wim.taymans@gmail.com>
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	* gst/audioconvert/audioconvert.c:
	Include math.h to fix compilation.

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2007-06-29  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
	Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
	format, as produced by some dc1394 cameras like the iSight.
	See http://www.fourcc.org/yuv.php#IYU1

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

	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	(check_default), (audio_convert_prepare_context),
	(audio_convert_clean_context), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_dithering_get_type),
	(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
	(gst_audio_convert_init), (gst_audio_convert_set_caps),
	(gst_audio_convert_set_property), (gst_audio_convert_get_property):
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	(gst_audio_quantize_setup_noise_shaping),
	(gst_audio_quantize_free_noise_shaping),
	(gst_audio_quantize_setup_dither),
	(gst_audio_quantize_free_dither),
	(gst_audio_quantize_setup_quantize_func),
	(gst_audio_quantize_setup), (gst_audio_quantize_free):
	* gst/audioconvert/gstaudioquantize.h:
	Implement dithering and noise shaping in audioconvert. By default now
	TPDF dithering (and no noise shaping) will be used when converting
	from a higher bit depth to 20 bit depth or smaller, otherwise
	everything will be as it is now.
	For the last audioconvert in a pipeline it would make sense to
	use some kind of noise shaping, enabling it by default for all
	conversions would give undesired results though. Fixes #360246.
	* tests/check/elements/audioconvert.c: (setup_audioconvert),
	(GST_START_TEST):
	Adjust unit test for the new audioconvert.

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2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
	Use other metrics as well when estimating the buffer level.

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2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
	Small debug improvement.

	* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
	(plugin_init):
	Tweak the rate estimation period.
	When calculating the buffer filledness in rate estimation mode, don't
	mix it with other metrics.

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2007-06-28  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
	(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
	When creating the groups, allow for a 5 second, unlimited buffers
	preroll phase after which we expose the group.
	When the group is exposed, use a small number of buffers up to a 2
	second limit. Also disconnect the overrun signal from multiqueue when we
	exposed the group because it is not needed anymore.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
	  to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
	  (#451707); also, output some debugging info when dealing with
	  freeform strings.

	* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
	  Add unit test for the above.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
	  Add description for Windows Media RTP caps.

	* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
	  Remove RTP fields that don't define the format from caps.

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2007-06-27  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	  Skip empty buffers, but not empty header buffers. That way the original
	  vorbisdec unit test still passes (#451145); also, take into account
	  that those empty packets might carry a granulepos.

	* tests/check/Makefile.am:
	* tests/check/elements/vorbisdec.c:
	(_create_codebook_header_buffer), (_create_audio_buffer),
	(GST_START_TEST), (vorbisdec_suite):
	  Add unit test that sends an empty packet.

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

	* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	Don't error out on 0-sized packets, just emit a warning because this is
	not a fatal error. Fixes #451145.

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

	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.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 (#450875).

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2007-06-23  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
	The chain should be freed if we error out here, else it will leak.
	* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
	(cleanup_decodebin):
	Don't forget to *properly* remove the signals, else it will leak.

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

	* MAINTAINERS:
	Updating all the maintainers files

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2007-06-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
	(main):
	  Destroy and recreate parse-launch based pipeline after stop to be able
	  to play again. Reorder some code and add more comments.

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2007-06-20  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin2.c: (analyze_new_pad):
	When handling a delayed-caps notification case, mark
	the group as dynamic so that the nbdynamic count is
	incremented and decremented correctly. Fixes: #449156
	Patch by: Wim Taymans <wim@fluendo.com>

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2007-06-19  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_init): Enable pull-mode operation.

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2007-06-19  Michael Smith <msmith@fluendo.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Change minimum rate back to 1000 to allow low-sample-rate wav files
	  to play back.

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2007-06-17  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/vi.po:
	  Update translations.

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2007-06-15  David Schleef  <ds@schleef.org>

	* gst/playback/gstqueue2.c:
	  Fix compile error from ignored return value.

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2007-06-15  Michael Smith <msmith@fluendo.com>

	* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
	  Update tmpbuf for all neccesary rows, not just one, as is required
	  when downscaling.
	  Fixes #402076.

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2007-06-15  Michael Smith <msmith@fluendo.com>

	* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
	(eos_buffer_probe):
	  Add a test that ensures we set DELTA_UNIT on all non-header,
	  non-video buffers, if we have a video stream.
	* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
	(gst_ogg_mux_process_best_pad):
	  Move setting delta_pad to earlier, where we inspect all pads, so
	  that leading audio pages don't get DELTA_UNIT unset if they come
	  before the first DELTA_UNIT from video pages. Fixes the newly-added
	  test. Fixes #385527.

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2007-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/streamheader.c: (streamheader_suite):
	  Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
	  fails on the p5-ppc64 build bot and the failure looks like it is due
	  to the same issue as #348114, ie. a compiler bug.

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2007-06-13  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_create_read):
	Fix build on MacOSX.

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

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
	Fix compilation on mingw. Fixes #446972.

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2007-06-12  Wim Taymans  <wim@fluendo.com>

	Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (update_buffering),
	(gst_queue_locked_enqueue):
	Fix a division by zero when the max percent is <= 0. Fixes #446572.
	also update the buffering status when receiving events. Fixes #446551.

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

	Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_peer_query),
	(gst_queue_handle_src_query):
	Wait for preroll before attempting to forward a duration query upstream.
	Fixes #445505.

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2007-06-07  Sebastien Moutte  <sebastien@moutte.net>

	* gst-libs/gst/rtp/gstbasertpdepayload.c: 
	(gst_base_rtp_depayload_set_gst_timestamp):
	Use G_GINT64_CONSTANT macro for int64 constant.
	* win32/common/libgstinterfaces.def:
	* win32/common/libgsttag.def:
	Add new exported functions.

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2007-06-07  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
	  The BOS page of the first Dirac video stream needs to come before
	  the BOS page of any Vorbis streams or other audio streams, just like
	  it is with Theora.

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2007-06-07  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_get_range):
	Fix compilation.

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

	Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_init),
	(gst_queue_handle_sink_event), (gst_queue_chain),
	(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
	(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
	(gst_queue_src_activate_pull):
	Add pull based scheduling and fix some deadlocks. Fixes #444523.
	Does not yet completely work because duration queries upstream won't
	block yet.

Wim Taymans's avatar
Wim Taymans committed
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2007-06-06  Wim Taymans  <wim@fluendo.com>

	* configure.ac:
	* gst/playback/gstqueue2.c: (gst_queue_create_read):
	Some more fseeko checks.

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

	* configure.ac:
	check for large file support.

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

	Based on a patch by Sven Arvidsson <sa at whiz dot se>:

	* gst/subparse/gstsubparse.c: (parse_subrip),
	(subviewer_unescape_newlines), (parse_subviewer),
	(gst_sub_parse_data_format_autodetect),
	(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
	* gst/subparse/gstsubparse.h:
	Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	Add a unit test for both SubViewer formats.

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2007-06-01  Michael Smith <msmith@fluendo.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	  Don't overflow intermediate values when seeking to large time values
	  in audiotestsrc.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstqueue2.c: (gst_queue_have_data),
	(gst_queue_create_read), (gst_queue_read_item_from_file),
	(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
	Include stdio to define fseeko.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

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

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
	(gst_v4lsrc_query):
	Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.

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

	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  our own implementation.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state):
	Handle timestamp wraparound.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gsturidecodebin.c: (no_more_pads_full),
	(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
	(gst_uri_decode_bin_change_state):
	Make sure we name srcpads uniquely even when using different internal
	decodebins.
	Signal no-more-pads when no more dynamic elements exist.
	Remove pads on cleanup.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>

	* gst/playback/gstqueue2.c: (gst_queue_class_init),
	(gst_queue_init), (gst_queue_finalize),
	(gst_queue_write_buffer_to_file), (gst_queue_have_data),
	(gst_queue_create_read), (gst_queue_read_item_from_file),
	(gst_queue_open_temp_location_file),
	(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_is_empty), (gst_queue_is_filled),
	(gst_queue_change_state), (gst_queue_set_temp_location),
	(gst_queue_set_property):
	Add support for filebased buffering. Fixes #441264.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
	(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
	(caps_notify_group_cb), (gst_decode_group_new),
	(gst_decode_group_free):
	Add support for delayed caps fixation when autoplugging.
	Optimize cases where a multiqueue is not needed/wanted, like right after
	anything that is not a demuxer.

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
	consideratly speedup ogg chain detection by not trying to find a base
	timestamp for skeleton streams. 

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
	(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_remove_flush),
	(gst_multi_fd_sink_remove_client_link),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_handle_clients):
	* gst/tcp/gstmultifdsink.h:
609
	Add support for remove_flush.
610

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2007-06-05  Wim Taymans  <wim@fluendo.com>

	* docs/design/draft-keyframe-force.txt:
	* ext/theora/theoraenc.c: (theora_enc_sink_event),
	(theora_enc_chain):
	Add draft design for forcing keyframes in encoders and implement in
	theoraenc.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-05  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	  Back to CVS

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=== release 0.10.13 ===

2007-06-05  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.13, "What's Going on?"

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2007-05-31  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	In riff, the depth is stored in the size field but it just means that
	the least significant bits are cleared. We can therefore just play
	the sample as if it had a depth == width. Fixes: #440997

	Patch by: Wim Taymans <wim@fluendo.com> 
	Patch by: Sebastian Dröge  <slomo@circular-chaos.org>

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2007-05-31  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/floatcast/floatcast.h:
	Define inline when needed on win32 builds. Fixes: #441295
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	Patch by: Sebastien Moutte  <sebastien@moutte.net>
646

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2007-05-29  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (queue_overrun),
	(no_more_pads_full):
	Stop buffering when the group is commited because the queues filled up.
	Fixes #442024.

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

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	(gst_alsa_mixer_free), (gst_alsa_mixer_update),
	(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
	(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
	(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_interface_supported),
	(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
	(gst_alsa_mixer_element_set_property),
	(gst_alsa_mixer_element_get_property),
	(gst_alsa_mixer_element_change_state):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
	* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
	(gst_mixer_option_changed):
	* gst-libs/gst/interfaces/mixer.h:
	Revert commits towards #152864 made so far. We'll pick it up again
	after the 0.10.13 release.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	After an interrupt (PAUSED/flush) assume that the next sample should not
	be aligned to the previous sample. Fixes #417992.

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Don't add channels and rate fields to the template caps for
	  audio/x-dts, as wavparse might not always be able to set them,
	  which would then lead to 'caps are not a real subset of the
	  template caps' warnings.

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2007-05-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
	Handle unknown or invalid pads without crashing, as might occur if
	a media file like an mp3 is specified as a subtitle file.
	Fixes: #410039

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2007-05-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
	(setup_sinks):
	Block the subtitle bin output queue before ghosting it and linking,
	then unblock after. This avoids spurious not-linked errors caused 
	by the queue starting up (because it gets linked when it is ghosted). 
	Fixes: #350299

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2007-05-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
	Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
	file. Avoids flukes where the input gets typefound to some valid but
	useless type.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
	(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
	  Add unit test for gnomevfssink seeking and position reporting for
	  file:// URIs.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
	(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
	(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
	* ext/gnomevfs/gstgnomevfssink.h:
	  Fix position reporting, especially after a seek (from upstream),
	  see #412648.

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2007-05-22  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cdparanoia/gstcdparanoiasrc.c:
	  Repair umlaut.

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2007-05-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Specify the full valid range for MP3 samplerates. Fixes a regression
	caused by extra header checks since the last release.

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2007-05-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	Fix a locking-order bug I introduced with my changes the other day.
	Patch by Mike Smith.

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2007-05-21  Michael Smith <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_data_packet):
	  Don't look inside 0-length packets (which indicate duplicated
	  frames)

Wim Taymans's avatar
Wim Taymans committed
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2007-05-21  Wim Taymans  <wim@fluendo.com>

	* ext/cdparanoia/gstcdparanoiasrc.c:
	(gst_cd_paranoia_src_read_sector):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	Small cleanups.

	* ext/theora/theoradec.c: (theora_dec_sink_event):
	Fix typo.

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_set_gst_timestamp):
	Add some FIXME

	* gst/playback/gstdecodebin.c: (queue_underrun_cb):
	And some debug info when a FIXME path is hit.

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

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init),
	(gst_base_rtp_audio_payload_init),
	(gst_base_rtp_audio_payload_finalize),
	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
	(gst_base_rtp_payload_audio_handle_event):
	Some cleanups, remove minptime property as it is now in the parent
	class.
	Override parent class event function.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_class_init), (gst_basertppayload_init),
	(gst_basertppayload_event), (gst_basertppayload_set_property),
	(gst_basertppayload_get_property):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Add min-ptime property.
	Add handle-event vmethod. Fixes #415001.

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2007-05-18  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_change_state):
	  Fix typo in comment.

	* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
	  free_dynamics, pad_probe, close_pad_link, try_to_link_1,
	  get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
	  close_link):
	* gst/playback/gstplaybin.c (gst_play_bin_set_property,
	  gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
	  Remove trailing whitespaces in comments.

	* gst/volume/Makefile.am:
	  Fix tabs.

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2007-05-18  Stefan Kost  <ensonic@users.sf.net>

	patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
	  set_option, get_option, _gst_reserved):
	  Revert reordering functions (keep ABI).

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2007-05-17  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
	(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
	(gst_ximagesink_show_frame):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_show_frame):
	When we create our own window, indicate that we handle the 
	WM_DELETE client message from the window manager, so that it won't 
	kill our window (and our app) along with it. Handle ClientMessage,
	post an error on the bus, and close the window. Further buffers
	arriving will result in a FlowError because the window has been
	destroyed.

	Fixes: #393975

	Clean up the X event handling loop and make them the same for
	both xvimagesink and ximagesink while I'm at it.

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

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
	Make decodebin2 autoplug depayloaders too.

	* gst/playback/gsturidecodebin.c: (source_new_pad):
	Set the newly created decoder in a usable state when autoplugging a
	dynamic source such as RTSP.

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2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c: (cb_probe):
	  Ignore video-codec tag for audio streams and ignore audio-codec tags
	  for video streams. Should make codec name collection a bit more
	  robust against sloppy demuxers that send tag events containing both
	  tags down each pad.

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

	* gst/playback/gstqueue2.c: (update_rates):
	Tweak the buffering thresholds a little.
	Update the buffer size with the previously calculate rate instead of
	only when we calculate a new rate so that we get smoother buffering
	updates.

	* gst/playback/Makefile.am:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
	(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
	(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
	(gst_uri_decode_bin_get_property), (unknown_type),
	(add_element_stream), (no_more_pads_full), (no_more_pads),
	(source_no_more_pads), (new_decoded_pad), (array_has_value),
	(gen_source_element), (has_all_raw_caps), (analyse_source),
	(remove_decoders), (make_decoder), (remove_source),
	(source_new_pad), (setup_source), (decoder_query_init),
	(decoder_query_duration_fold), (decoder_query_duration_done),
	(decoder_query_position_fold), (decoder_query_position_done),
	(decoder_query_latency_fold), (decoder_query_latency_done),
	(decoder_query_seeking_fold), (decoder_query_seeking_done),
	(decoder_query_generic_fold), (gst_uri_decode_bin_query),
	(gst_uri_decode_bin_change_state), (plugin_init):
	New element that intergrates a source, optional buffering element and
	decodebin.

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2007-05-17  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Bump libtheora requirement to 1.0alpha5 for the pixformat check
	  (also has a .pc file, so we don't need the fallback check any
	  longer). Fixes #438840.

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

	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
	(apply_segment), (apply_buffer), (update_buffering),
	(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_filled),
	(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
	(plugin_init):
	fix build.

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

	* gst/playback/Makefile.am:
	* gst/playback/gstqueue2.c: (gst_queue_get_type),
	(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
	(gst_queue_getcaps), (gst_queue_bufferalloc),
	(gst_queue_acceptcaps), (update_time_level), (apply_segment),
	(apply_buffer), (update_buffering), (reset_rate_timer),
	(update_rates), (gst_queue_locked_flush),
	(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	(gst_queue_handle_sink_event), (gst_queue_is_empty),
	(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
	(gst_queue_loop), (gst_queue_handle_src_event),
	(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
	(gst_queue_src_activate_push), (gst_queue_change_state),
	(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
	On our way to playbin2 this is the new network queue that does buffering
	all by itself using high and low watermarks. It can also measure up and
	downstream bandwidth to optimally size the queue.

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2007-05-17  Michael Smith <msmith@fluendo.com>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
	  Use the segment->last_stop value to calculate the next timestamp to
	  generate after a seek; not the segment->start value.

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2007-05-15  David Schleef  <ds@schleef.org>

	* docs/Makefile.am: Install docs even when --disable-gtk-doc
	  is disabled.  This matches the behavior of gtk+.  Fixes #349099.

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2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
	Some more chained streaming ogg timestamp fixes.

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2007-05-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_handle_page):
	Add some FIXMEs.
	Fix chain start/stop segment handling based on patch by
	<ahalda at cs dot mcgill dot ca> see #320984.

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2007-05-15  Michael Smith <msmith@fluendo.com>

	* configure.ac:
	  We don't require a C++ compiler. So don't require one.

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

	* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track):
	  Apply some of the cleanup Tim suggested in #152864 afterwards.

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

	patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
	  _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
	  gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
	  gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
	  gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
	  gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
	* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
	* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
	  gst_alsa_mixer_element_interface_supported,
	  gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
	  gst_alsa_mixer_element_set_property,
	  gst_alsa_mixer_element_get_property,
	  gst_alsa_mixer_element_change_state):
	* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
	* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
	  gst_mixer_option_changed):
	* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
	  volume_changed, option_changed, _gst_reserved):
	  Implement notification for alsamixer. Fixes #152864

996 997 998 999 1000 1001
2007-05-14  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add support for video/x-raw-bayer.

1002 1003 1004 1005 1006 1007
2007-05-12  David Schleef  <ds@schleef.org>

	* sys/xvimage/xvimagesink.c:
	  Add some sanity checking for the XVImage size returned by X.
	  Related to #377400.

1008 1009 1010 1011 1012 1013 1014 1015
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Parse and use additional caps fields as described in updated
	application/x-rtp caps spec.

1016 1017 1018 1019 1020 1021 1022
2007-05-12  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	(gst_ogg_demux_collect_chain_info):
	If there is a stream in a chain without any data packets, ignore the
	stream in the total length calculations. Might be related to #436820.

1023 1024 1025 1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043
2007-05-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
	(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
	(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
	(mpeg_video_type_find), (mpeg_video_stream_type_find),
	(plugin_init):

	Consolidate and re-work our mpeg system stream detection to probe
	more packets and produce a higher confidence result. Fixes a
	regression caused by lowering the typefind probability last year
	- related to bug #397810. Remove the redundant MPEG-1 specific 
	typefind function, as the new one detects both MPEG-1 & MPEG-2
	happily.

	Also cleanup the MPEG elementary and MPEG-TS detection functions a
	little. 

	Tested against my media test directory, with some improvements and
	no regressions.

1044 1045 1046 1047 1048 1049 1050
2007-05-10  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
	(queue_out_of_data):
	Connect to the new queue "pushing" signal instead of the broken
	"running" one.

1051 1052 1053 1054 1055 1056 1057 1058 1059 1060
2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	Move variable declaration before the first instruction.
	* gst/videotestsrc/videotestsrc.c:
	Define M_PI if it's not defined yet.
	* win32/common/libgstrtp.def:
	Add new exported functions.

1061 1062 1063 1064 1065
2007-05-09  Michael Smith <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  gst_pad_push_event() does not return a GstFlowReturn!

1066 1067 1068 1069 1070 1071
2007-05-09  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/scrubby.c: (stop_cb), (main):
	* tests/examples/seek/seek.c: (do_seek):
	Some small cosmetic changes.

1072 1073 1074 1075 1076 1077 1078 1079
2007-05-08  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
	  gst_adder_change_state):
	* gst/adder/gstadder.h (bps, offset, collect_event, segment,
	  segment_pending, segment_position, segment_rate):
	  Handle playback-rate on adder.

1080 1081 1082 1083 1084 1085 1086 1087 1088 1089
2007-05-07  Michael Smith <msmith@fluendo.com>

	* ext/theora/gsttheoradec.h:
	* ext/theora/theoradec.c: (gst_theora_dec_reset),
	(theora_dec_sink_event), (theora_handle_comment_packet),
	(theora_handle_type_packet), (theora_dec_change_state):
	  Don't push events (newsegment, tags) before initialising the
	  decoder.
	  This is neccesary for seeking to work correctly in gnonlin.

1090 1091 1092 1093 1094 1095 1096 1097 1098 1099 1100 1101 1102 1103
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst/adder/gstadder.c:
	* gst/audiotestsrc/gstaudiotestsrc.c
	  (gst_audio_test_src_create_white_noise):
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
	  VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
	  volume_sink_template, volume_src_template, gst_volume_init,
	  volume_process_double, volume_process_int16,
	  volume_process_int16_clamp):
	  Doc fixes and formatting.

1104 1105 1106 1107 1108 1109 1110
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  Minimal check for volume's GstController usability; also another
	  test for #422295.

1111 1112 1113 1114 1115 1116 1117
2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix it so that it (a) makes sense and (b) doesn't break
	  everything cdda-related including the unit test.

1118 1119 1120 1121 1122 1123
2007-05-04  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/cdda/gstcddabasesrc.c:
	(gst_cdda_base_src_add_track):
	  Fix build when disabling asserts.

1124 1125 1126 1127 1128 1129 1130 1131 1132 1133 1134 1135 1136
2007-05-03  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
	  When XShm is not available, we might get row strides that are not
	  rounded up to multiples of four; this is bad, because virtually
	  every RGB-processing element in GStreamer assumes rowstrides are
	  rounded up to multiples of four, so let's allocate at least enough
	  memory to avoid crashes in this case. The image will still be
	  displayed distorted though if this happens, so that still needs
	  fixing (maybe by allocating a bigger image with an 'even' width
	  and then clipping it appropriately when rendering - something for
	  Xlib aficionados in any case).

1137 1138 1139 1140 1141 1142
2007-05-03  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If a buffer doesn't have a timestamp, assume it's contiguous with
	  the previous buffer, and synthesise timestamps appropriately.

1143 1144 1145 1146 1147 1148
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* tests/check/elements/videorate.c: (GST_START_TEST):
	Set buffer timestamp to a valid value in order to test the buffer
	really does stay in videorate.

1149 1150 1151 1152 1153 1154
2007-05-03  Edward Hervey  <edward@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	There is no sensible way to handle incoming buffers which don't have a
	valid timestamp. We therefore discard them and wait for the next one.

1155 1156 1157 1158 1159 1160
2007-05-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
	* gst/playback/gstdecodebin2.c: (plugin_init):
	  Better error message for text files.

1161 1162 1163 1164 1165
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
	Fix offset bug in generation RR packets.

1166 1167 1168 1169 1170 1171 1172 1173 1174
2007-04-27  Julien MOUTTE  <julien@moutte.net>

	* ext/theora/theoradec.c: (_theora_granule_time),
	(theora_dec_push_forward), (theora_handle_data_packet),
	(theora_dec_decode_buffer): Calculate buffer duration correctly
	to generate a perfect stream (#433888).
	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont): Glib provides ABS.

1175 1176 1177 1178 1179 1180 1181 1182 1183 1184 1185
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix RB block parsing and writing.
	Add support for constructing BYE packets.

1186 1187 1188 1189 1190 1191 1192 1193 1194
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
	(gst_base_audio_src_create):
	* po/POTFILES.in:
	  When posting a warning message because samples were dropped, post
	  something more intelligible than he default error message for clock
	  errors which is just confusing in this context (#432984).

1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208 1209 1210 1211
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
	(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
	(read_packet_header), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
	(gst_rtcp_packet_sdes_get_item_count),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_entry),
	(gst_rtcp_packet_sdes_next_entry),
	(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
	(gst_rtcp_packet_sdes_add_entry):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Implement code to write SR, RR and SDES packets.

1212 1213 1214 1215 1216 1217 1218
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>

	* sys/ximage/ximagesink.c:
	  Fix build if XShm is not available (#432362).

1219 1220 1221 1222 1223 1224 1225
2007-04-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
	Initalize the AudioConvertCtx with zeroes, otherwise it will contain
	pointers to random memory which are passed to g_free() when
	audio_convert_prepare_context() is called the first time.

1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237
2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams <dcbw redhat com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  Don't leak incoming buffer if gst_pad_push() returns a
	  non-OK flow. Fixes #432755.
	 
	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	  Unit test for the above by Yours Truly.

1238 1239 1240 1241 1242 1243
2007-04-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
	(gst_adder_sink_event), (gst_adder_collected):
	  Fix non-flushing segmented seeks, Fixes #340060 for me

1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Olivier Crete  <tester at tester ca>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init),
	(gst_base_rtp_audio_payload_init),
	(gst_base_rtp_audio_payload_dispose):
	  Chain up to parent class in dispose function; get rid of
	  unnecessary 'diposed' flag in private structure (#415001).

1255 1256 1257 1258 1259 1260 1261 1262 1263
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs.types:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_class_init):
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	  Some minor docs fixes and additions; also add missing 'Since' bits.

1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Zeeshan Ali  <zeenix gmail com>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_handle_frame_based_buffer),
	(gst_base_rtp_audio_payload_handle_sample_based_buffer),
	(gst_base_rtp_audio_payload_push):
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  The recently-added gst_base_rtp_audio_payload_push() should take an
	  object of type GstBaseRTPAudioPayload as first argument (#431672).

1276 1277 1278 1279 1280
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioresample/gstaudioresample.c:
	  Make more functions static, just because we can.

1281 1282 1283 1284 1285
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/audioresample.c:
	  Add unit test for audioresample shutdown crasher (#420106).

1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299 1300 1301
2007-04-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/subparse/gstsubparse.c:
	* gst/subparse/samiparse.c:
	  Use GST_DISABLE_XML here

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
	(gst_xvimagesink_buffer_alloc),
	(gst_xvimagesink_navigation_send_event):
	* sys/xvimage/xvimagesink.h:
	  Include stdlib.h when using atoi.
	  
	* tests/check/elements/playbin.c: (playbin_suite):
	  Use GST_DISABLE_REGISTRY here

1302 1303 1304 1305 1306 1307 1308 1309
2007-04-19  Michael Smith  <msmith@fluendo.com>

	* ext/theora/gsttheoraenc.h:
	* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
	(theora_enc_sink_event), (theora_enc_change_state):
	  Track initialisation state; don't try to use encoder state if we're
	  not initialised (it'll segfault).

1310 1311 1312 1313 1314
2007-04-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/.cvsignore:
	Fix build.

1315 1316 1317 1318 1319
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Allow random depths between 1 and 32 instead of only multiplies of 8.

1320 1321 1322 1323 1324 1325
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Set the maximum number of channels for PCM and float in the correct
	place to have it also used when creating the template caps.

1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338 1339 1340 1341 1342 1343
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Correctly support 4, 6 and 8 channels with normal PCM and float
	wav files.

	Fix the depth and signedness calculation in extensible wav files and
	also handle 1, 2, 4, 6, 8 channels here when a file without channel
	mask is found.

	Add support for float, alaw and mulaw in extensible wav files.

	This allows correct playback of all but 5 files from
	http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
	
	(gst_riff_create_audio_template_caps):
	Add voxware and float formats to the template caps.	

1344 1345 1346 1347 1348 1349 1350 1351 1352 1353 1354
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
	Fix unused variable warning if HAVE_LOCALTIME_R is undefinied

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
	Use the correct format strings for integer formats.

1355 1356 1357 1358 1359 1360 1361
2007-04-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
	  Don't use pad_alloc_buffer_and_set_caps to create a small header
	  packet, or, worse, to create a big temporary video buffer using the
	  src pad.

1362 1363 1364 1365 1366 1367 1368
2007-04-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, buffer_probe_cb, GST_START_TEST):
	  Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.

1369 1370 1371 1372 1373 1374 1375
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
	  streamheader_suite):
	  Add another test set up for failure

1376 1377 1378 1379 1380 1381 1382
2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

	* tests/check/Makefile.am:
	* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
	  GST_START_TEST, streamheader_suite, main):
	  Add a test for the streamheader bug Wim fixed.

1383 1384 1385 1386 1387
2007-04-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/theora/theoradec.c: (theora_dec_sink_event):
	Fix misleading comment.

1388 1389 1390 1391 1392
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  More sanity checks for the header fields.

1393 1394 1395 1396 1397 1398
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Try encodings from all environment variables, not just those in the
	  first environment variable that is set.

1399 1400 1401 1402 1403 1404 1405 1406 1407 1408
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_chain):
	Add some debug.

	* tests/check/elements/videorate.c: (GST_START_TEST),
	(videorate_suite):
	Added check for videorate changing caps handling. Closes #421834.

1409 1410 1411 1412 1413 1414
2007-04-12  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  Use scale functions to avoid overflow when calculating duration of 
	  vorbis buffers.

1415 1416 1417 1418 1419 1420 1421 1422 1423 1424
2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  API: add gst_tag_freeform_string_to_utf8() (#405072).

	* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
	  Use gst_tag_freeform_string_to_utf8() here.

1425 1426 1427 1428 1429 1430 1431 1432 1433
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
	(gst_gdp_pay_sink_event):
	Make sure we set the IN_CAPS flag correctly.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Get the IN_CAPS flag before we call functions that mess with the flags.

1434 1435 1436 1437 1438 1439 1440 1441 1442
2007-04-10  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
	  gst_gdp_pay_chain, gst_gdp_pay_sink_event):
	  Only stamp buffers with offset/offset_end right before they get
	  pushed.  This ensures offset continuity, which was not the case
	  before as shown by
	  gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE

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

	* gst/playback/gstplaybin.c: (add_sink),
	(gst_play_bin_change_state):
	Activate sync in playbin, we are ready to handle it for live streams.

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2007-04-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/playbin.c:
	(test_sink_usage_video_only_stream), (playbin_suite):
	  Add small test for stream-info-value-array code paths.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving):
	Don't try to create invalid calibration parameters by making the
	internal time go backwards, instead make external time go forward.

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

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

	* gst/playback/gstplaybasebin.c: (add_stream):
	Fix leak in add_stream(), when g_value_set_object() increases the
	refcount of streaminfo object. Fixes #426250.

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2007-04-03  David Schleef  <ds@schleef.org>

	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c:
	* gst/videotestsrc/videotestsrc.h:
	  Add a test pattern called "circular", which has concentric
	  rings with varying radial frequency.  The main purpose of this
	  pattern is to test fidelity loss in a filter or scaler element.
	  Notably, this pattern is scale invariant, and is optimally viewed
	  with a width (and height) of 400.

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

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

	* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
	(deactivate_free_recursive):
	Decodebin2 doesn't unref pads it obtains in some occasions:
	- multiqueue src pads, when either connecting further or exposing
	- sink pads of new autoplugged elements
	- peer pads when recursively freeing elements
	Fixes #425455.

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	Add audio/x-raw-float support, now that audioconvert support
	non-native endianness floats.

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2007-03-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	  gstreamer-plugins-base.pc doesn't exist, it's
	  gstreamer-plugins-base-0.10.pc.

René Stadler's avatar
René Stadler committed
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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: René Stadler <mail at renestadler dot de>
	with some minor changes

	* gst-libs/gst/floatcast/floatcast.h:
	Use more efficient float endianness conversion functions that don't
	involve 2 function calls per value.
	* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	(check_default), (audio_convert_prepare_context):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_parse_caps), (make_lossless_changes):
	Support non-native endianness floats as input and output.
	Fixes #339838.
	* tests/check/elements/audioconvert.c: (verify_convert),
	(GST_START_TEST):
	Add unit tests for the non-native endianness float conversions.

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

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_base_init),
	(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Add Private structure.
	Bring element code to 2007.
	Parse clock-base caps param and use it when generating the
	newsegment.
	Reset variables before going to PAUSED.
	Fix some docs.

Wim Taymans's avatar
Wim Taymans committed
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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_audio_payload_get_adapter):
	Add RTCP docs.
	Fix some more docs.

	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
	(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
	(gst_rtcp_buffer_get_packet_count), (read_packet_header),
	(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
	(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
	(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
	(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
	(gst_rtcp_packet_sr_get_sender_info),
	(gst_rtcp_packet_sr_set_sender_info),
	(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
	(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
	(gst_rtcp_packet_sdes_get_chunk_count),
	(gst_rtcp_packet_sdes_first_chunk),
	(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
	(gst_rtcp_packet_sdes_first_item),
	(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
	(gst_rtcp_packet_bye_get_ssrc_count),
	(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
	(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	(gst_rtcp_packet_bye_get_reason_len),
	(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Add new helper object for parsing and creating RTCP messages.

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	PCM samples with width=8 must be always unsigned, no matter what
	depth they have.

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2007-03-29  Andy Wingo  <wingo@pobox.com>

	* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
	perfect offsets also, not just timestamps.

	* tests/check/elements/videorate.c (test_more): Test that given
	any incoming offsets, that videorate produces perfect offsets.

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

	* gst-libs/gst/riff/riff-ids.h:
	Add some more RIFF formats.

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

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	(gst_rtp_buffer_default_clock_rate):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Fix fixed payload names and docs.
	Added method to get the default clock rates of fixed payload types.
	API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()

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2007-03-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

	* tests/check/pipelines/.cvsignore:
	Add new vorbisdec test to cvsignore.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
	(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
	(gst_base_audio_sink_set_property),
	(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
	(clock_convert_external), (gst_base_audio_sink_resample_slaving),
	(gst_base_audio_sink_skew_slaving),
	(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_async_play):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	Store private stuff in GstBaseAudioSinkPrivate.
	Add configurable clock slaving modes property.
	API:: GstBaseAudioSink::slave-method property
	Some more latency reporting tweaks.
	Added skew based clock slaving correction and make it the default until
	the resampling method is more robust.

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2007-03-27  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioconvert/audioconvert.c:
	Add docs to the integer pack functions and implement proper
	rounding. Before we had rounding towards negative infinity, i.e.
	always the smaller number was taken. Now we use natural rounding,
	i.e. rounding to the nearest integer and to the one with the largest
	absolute value for X.5. The old rounding introduced some minor
	distortions. Fixes #420079
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix one unit test that assumed the old rounding and added unit tests
	for checking signed/unsigned int16 <-> signed/unsigned int16 with
	depth 8, one for signed int16 <-> unsigned int16 and one for the new
	rounding from signed int32 to signed/unsigned int16.

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2007-03-27  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
	(gst_audio_convert_transform_caps):
	  Fix typo in debug line introduced recently, as pointed out on irc.

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2007-03-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	* tests/check/libs/tag.c: (GST_START_TEST):
	  Make sure we parse floating-point numbers in vorbis comments
	  correctly with either '.' or ',' as separator, no matter what
	  the current locale is. Add unit test for this too.

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2007-03-26  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler  <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
	  When writing out floating-point numbers to vorbis comment tags, always
	  use the same character as separator no matter what the current locale is
	  (fixes #423051).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit tests for replaygain tags in vorbis comments (closes #423055).

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2007-03-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
	  vorbis_handle_data_packet):
	  Correctly set DURATION to generate a timestamp-continuous stream.
	  One bug left at the end; see
	  ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
	* tests/check/Makefile.am:
	* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
	  Add a test to check this.  Without the above patch this test fails.

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2007-03-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/rtp/Makefile.am:
	The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.

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2007-03-23  Michael Smith  <msmith@fluendo.com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	(gst_video_rate_reset), (gst_video_rate_chain):
	  If videorate changes caps, we can no longer use the old buffer
	  (which may have a different size, incompatible with our caps).
	  So don't do that; just duplicate the new frame more times.

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2007-03-22  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	Remove playbin's override of the set_clock vmethod. It's irrelevant
	after Wim's commit on the 19th.

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

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
	(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
	* ext/gnomevfs/gstgnomevfssrc.h:
	Don't cache file sizes. Fixes #341078.

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

	* gst/playback/gstplaybin.c: (add_sink):
	  Use GST_PTR_FORMAT to log caps. 

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

	Patch by: Young-Ho Cha <ganadist at chollian net>

	* gst/subparse/samiparse.c: (handle_start_font):
	  Special-case some more colour names that pango doesn't handle by
	  default. Fixes #420578.

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2007-03-20  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  If we get a zero-sized input buffer, don't pass it to libvorbis, as
	  that marks EOS internally. After that, libvorbis will buffer all
	  input data, and encode none of it, eventually leading to memory
	  exhaustion.

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2007-03-19  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (remove_fakesink):
	Don't post STATE_DIRTY anymore.

	* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
	(gst_play_bin_change_state):
	Remove stream_time reset in seek handling, core does that now.
	Disable clocking for live pipelines by forcing a NULL clock to the
	complete pipeline, core is too smart now for our previous hack.
	We can always autoplug in PAUSED now.

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

	* REQUIREMENTS:  Update this file, change the formatting to make
	it more consistent, plus more machine readable.

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2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(strip_width_64), (append_with_other_format):
	  Previous fix was too simplistic, and broke the tests. Use a better
	  approach; only strip 64 from widths for integer audio.

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2007-03-16  Michael Smith  <msmith@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	(gst_audio_convert_transform_caps):
	  We don't support 64 bit integer audio, so don't try to claim we can.
	  Stops us producing caps don't match our template caps.
	  Update comments.

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2007-03-15  Michael Smith  <msmith@fluendo.com>

	* gst/audioresample/gstaudioresample.c:
	(audioresample_check_discont), (audioresample_transform):
	  Don't trigger discontinuities for very small imperfections; a filter
	  flush will sound bad, and many plugins have rounding errors leading
	  to these.

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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 

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	Patch by Olivier Crete <olivier.crete@collabora.co.uk>

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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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	API: add "min-ptime" property to RTP base audio payloader.
	API: add gst_base_rtp_audio_payload_push().
	API: add gst_base_rtp_audio_payload_get_adapter().
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	Fixes #415001
	Indentation/whitespace/documentation fixes.

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2007-03-14  Julien MOUTTE  <julien@moutte.net>

	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	(audioresample_transform_size), (audioresample_do_output),
	(audioresample_transform), (audioresample_pushthrough): Handle
	discontinuous streams.
	* gst/audioresample/gstaudioresample.h:
	* tests/check/elements/audioresample.c:
	(test_discont_stream_instance), (GST_START_TEST),
	(audioresample_suite): Add a test for discontinuous streams.
	* win32/common/config.h: Updated.

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

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

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

	* gst/audioresample/debug.h:
	* gst/audioresample/resample.c: (resample_init):
	  Since I really am not interested in a debug line for each sample
	  being processed, move the library's debugging to its own category,
	  libaudioresample

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2007-03-13  Michael Smith  <msmith@fluendo.com>

	* ext/theora/theoradec.c: (theora_handle_type_packet):
	  Since the plugin doesn't support anything other than 4:2:0 right
	  now, post an error and fail if we get something else. Won't matter
	  until libtheora supports the other pixel formats, but hopefully
	  that'll be soon...

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
	Use gst_guint64_to_gdouble for conversion.
	* win32/MANIFEST:
	Add new files to the win32 MANIFEST.
	* win32/common/libgstaudio.def:
	* win32/common/libgstpbutils.def:
	Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstplaybin.dsp:
	Change the link to libgstpbutils.lib.
	* win32/vs6/libgstdecodebin2.dsp:
	Add a new project for decodebin2.
	* win32/vs6/libgstpbutils.dsp:
	Add a new project for pbutils.

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Also accept partial dates with only year and month,
	  like 1999-12-00 (fixes #410396 even more).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit test for the above.

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/subparse.c: (GST_START_TEST),
	(subparse_suite):
	  Add unit test for MPL2 subtitle format (#413799).

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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Kamil Pawlowski  <kamilpe gmail com>

	* gst/subparse/Makefile.am:
	* gst/subparse/gstsubparse.c:
	(gst_sub_parse_data_format_autodetect),
	(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
	(gst_subparse_type_find):
	* gst/subparse/gstsubparse.h:
	* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
	* gst/subparse/mpl2parse.h:
	  Add support for MPL2 subtitle format (#413799).

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

	* configure.ac:
	  We require core CVS for the new buffer metadata copy functions.

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

	* gst-libs/gst/tag/gstid3tag.c:
	Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.

Alex Lancaster's avatar
Alex Lancaster committed
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	Patch by: Alex Lancaster <alexl at users sourceforge net>

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

	* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
	(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
	Improve adapter usage and comments.

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

	* ext/pango/gsttextrender.c: (gst_text_render_chain):
	* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
	* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
	Use new metadata copy function.

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_transform):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
	Basetransform copied the metadata for us.

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

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
	(gst_text_overlay_video_event):
	  Some more logging. Only accept newsegment events in TIME format and
	  send a WARNING message if they are not in TIME format.

	* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
	(gst_sub_parse_chain), (gst_sub_parse_sink_event):
	* gst/subparse/gstsubparse.h:
	  No need to allocate GstSegment structure dynamically, just put it
	  into the instance structure; ignore newsegment events in BYTE
	  format and in particular don't let it overwrite our saved TIME
	  segment from the last seek.

1920 1921 1922 1923 1924 1925
2007-03-09  Michael Smith  <msmith@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
	  Replace AC3 typefinder with one that isn't terrible, and actually
	  works usefully.

1926 1927 1928 1929 1930 1931 1932
2007-03-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_transform):
	  fix error category and translatable string
	  

1933 1934 1935 1936 1937 1938
2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	  Fix up utils => pbutils here too.

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

	* gst/subparse/gstsubparse.c: (handle_buffer):
	  Break out of loop in chain function as soon as possible if we get
	  a non-OK flow return.

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

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Unref the mixer if the state change fails too (if the
	alsa devices are inaccessible, for example)

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

	* tests/check/Makefile.am:
	Don't test libvisual elements in the states check, because libvisual
	seems to leak internally.

	Re-enable the alsa and states tests now that there's new suppressions
	in gst.supp.

	* tests/check/elements/alsa.c: (GST_START_TEST):
	Don't leak the alsamixer we instantiated.

1963 1964 1965 1966 1967 1968 1969 1970 1971 1972 1973 1974 1975 1976 1977
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state), (gst_ximagesink_reset),
	(gst_ximagesink_finalize):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
	(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
	Move some cleanup stuff from the state change handler into a _reset()
	function that can be called from _finalize(). This ensures that things
	get freed even if (for some reason) the NULL->READY state transition
	fails in the parent class.
	Even if a parent state change fails, process our downward state change
	logic instead of bailing out early.
	Free the correct xcontext pointer in ximagesink's xcontext_clear.

1978 1979 1980 1981 1982 1983 1984 1985 1986 1987 1988 1989 1990 1991 1992 1993 1994
2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	Extra log line.

	* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
	* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
	Use pango_font_description_set_family_static instead of 
	pango_font_description_set_family to save a string copy (it was
	leaking due to the strdup anyway)

	* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
	* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
	* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
	Chain up in finalize.

1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008 2009
2007-03-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/interfaces/mixertrack.c:
	(gst_mixer_track_class_init), (gst_mixer_track_get_property),
	(gst_mixer_track_set_property):
	  API: add "untranslated-label" property which should be set by
	  implementations at construct time (#414645).

	* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Set "untranslated-label" when constructing mixer track objects.

	* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
	  Unit test to check the above.

2010 2011 2012 2013 2014
2007-03-07  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	Fix confusing debug message.

2015 2016 2017 2018 2019
2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-plugins-base.doap:
	update doap file with new version

Jan Schmidt's avatar
Jan Schmidt committed
2020 2021 2022 2023 2024
2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

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Jan Schmidt committed
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=== release 0.10.12 ===

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

	* configure.ac:
	  releasing 0.10.12, "Zombie Horde"

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

	* configure.ac:
	Bump version to 0.10.11.4 pre-release

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