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2006-09-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
	(gst_alsa_mixer_ensure_track_list):
	  Try harder to guess which mixer track is the master mixer
	  track (instead of just taking the first one that has a pvolume).
	  Fixes #342228.

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

	* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	(gst_audio_convert_transform_caps):
          Get structure-name just once.

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

	* tests/check/elements/audioresample.c: (GST_START_TEST):
	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	* tests/check/elements/volume.c: (GST_START_TEST):
	* tests/check/elements/vorbisdec.c: (GST_START_TEST):
	* tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
	(test_pipeline), (GST_START_TEST):
	* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
          Fix big batch of compiler warnings.

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

	* ext/gnomevfs/gstgnomevfssrc.c:
          Add docs about icydemux usage in connection with gnomevfssrc

	* ext/libvisual/visual.c:
	* ext/ogg/gstoggaviparse.c:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/ogg/gstoggparse.c:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst/audiorate/gstaudiorate.c:
	  More G_OBJECT macro fixing.

	* gst/audiotestsrc/gstaudiotestsrc.h:
          Fix wrong info in header due to copy & paste

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
	(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
	(gst_base_audio_src_create), (gst_base_audio_src_change_state):
	Do the delay calculation in the source/sink base classes as this is
	specific for the capture/playback mode.
	Try to fixate a bit better, like round depth up to a multiple of 8
	bigger than width.
	Handle underruns correctly by marking DISCONT on buffers and adjusting
	timestamps to handle the gap.
	Set offset/offset_end correctly on buffers.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
	(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	(gst_ring_buffer_read):
	Remove resync and underrun recovery from the ringbuffer.
	Fix ringbuffer read code on under/overrun.

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

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (fill_buffer), (check_queue),
	(queue_threshold_reached), (gst_play_base_bin_set_property),
	(gst_play_base_bin_get_property):
	* gst/playback/gstplaybasebin.h:
	Don't use a 0 low watermark when buffering, it is catching starvation
	way too late. Instead, use a 3 second queue with 30 and 95
	percent low/high watermarks. 
	Added queue-min-threshold property to configure low watermark.
	Use new _buffering message API.
	Make queue_threshold variable big enough to store a uint64 time value.
	API: playbin::queue-min-threshold property.

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

	* configure.ac:
	We require 0.10.10.1 now because of _wait_preroll().

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Use gst_base_sink_wait_preroll().

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

	* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
	* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
	Use DEBUG_OBJECT more.

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=== release 0.10.10 ===

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2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Michael Smith <msmith at fluendo dot com>

	* gst/tcp/gstmultifdsink.c: (is_sync_frame),
	(gst_multi_fd_sink_client_queue_buffer),
	(gst_multi_fd_sink_new_client):
	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(multifdsink_suite):
	  Fix implementation of sync-method 'next-keyframe'
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	  Closes #354594
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2006-09-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	patch by: Wim Taymans <wim at fluendo dot com>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	This patch removes the RANDOM flag that was incorrectly introduced with
	revision 1.91.  Fixes #354590

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2006-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Random variation in Makefile line to see if it makes the
	  gen64-base-full bot any happier.

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

	* tests/check/pipelines/oggmux.c: (oggmux_suite):
	  Disable test that fails at the moment (killed after timeout).

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

	Patch by: James Livingston  <doclivingston at gmail.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/.cvsignore:
	* tests/check/pipelines/oggmux.c: (get_page_codec),
	(check_chain_final_state), (fail_if_audio), (validate_ogg_page),
	(eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
	(test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
	(test_theora_vorbis), (oggmux_suite):
	  Add simple unit test for oggmux from #337026 with checking for the
	  EOS flags disabled for the time being.

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

	patch by: Alessandro Dessina <alessandro nnva org>

	* ext/ogg/gstoggmux.c:
	Add cmml caps to oggmux. Fixes #353912

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2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Returning a return value often helps. In this case, we
	  don't need the return value anyway, so just get rid of it.
	  Should make build bots much happier.

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2006-09-02  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
	(paint_get_structure), (gst_video_test_src_get_size),
	(gst_video_test_src_smpte), (gst_video_test_src_snow),
	(gst_video_test_src_unicolor), (paint_setup_AYUV),
	(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
	(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
	* gst/videotestsrc/videotestsrc.h:
	  Add support for AYUV and the various RGBA formats. Initialise
	  fields of paintinfo structs allocated on the stack.

	* tests/check/elements/videotestsrc.c: (right_shift_colour),
	(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
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	(check_rgb_buf), (videotestsrc_suite):
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	  Add unit tests for videotestsrc's RGB output.

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2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_pattern_get_type),
	(gst_video_test_src_set_pattern):
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
	(gst_video_test_src_black), (gst_video_test_src_white),
	(gst_video_test_src_red), (gst_video_test_src_green),
	(gst_video_test_src_blue):
	* gst/videotestsrc/videotestsrc.h:
	  Add more uni-colour patterns ("white", "red", "green", and "blue").

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2006-09-01  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
	  Fix stride for YVYU, should be word-aligned (#353658).

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2006-08-31  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/adder/gstadder.c: (gst_adder_src_event):
	  Fix build.

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2006-08-31  Edward Hervey  <edward@fluendo.com>

	* gst/adder/gstadder.c: (forward_event_func),
	(gst_adder_src_event), (gst_adder_collected),
	(gst_adder_change_state):
	* gst/adder/gstadder.h:
	Remember the start position asked in the incoming seeks, so we can
	output GST_EVENT_NEW_SEGMENT with a correct position value (instead
	of assuming it will always be 0).

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2006-08-31  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
	(gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_loop):
	Send the GST_EVENT_NEW_SEGMENT from the streaming thread.

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2006-08-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	  Return FALSE instead of returning a random false unit
	  size when the format isn't known/supported (even if
	  this shouldn't happen under normal circumstances).

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2006-08-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
	(gst_gnome_vfs_src_start):
	Try harder to get the size from a uri by using _info_uri() when
	_info_from_handle() does not give us enough info. 
	Also follow symlinks when getting the size.
	Partially Fixes #332864.

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2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Viktor Peters  <viktor dot peters at gmail dot com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
	(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
	(gst_alsa_mixer_set_record):
	* ext/alsa/gstalsamixertrack.c:
	(gst_alsa_mixer_track_update_alsa_capabilities),
	(alsa_track_has_cap), (gst_alsa_mixer_track_new),
	(gst_alsa_mixer_track_update):
	* ext/alsa/gstalsamixertrack.h:
	  Improve and fix mixer track handling, in particular better handling
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	  of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create
	  separate track objects for tracks that have both capture and playback
	  volume (and label them differently as well so they're not mistakenly
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	  assumed to be duplicates); classify mixer tracks that only affect
	  the audible volume of something (rather than the capture volume)
	  as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
	  for capture tracks to correspond to alsa-pswitch alsa-cswitch
	  (following the meaning documented in the mixer interface header
	  file); add support for alsa's exclusive cswitch groups; update/sync
	  state/flags better if mixer settings are changed by another
	  application. Fixes #336075.

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2006-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: add section about BUFFERING messages sent by playbin.

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2006-08-29  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain):
	  Ignore explicit DISCONT marked on buffers (which is often spurious,
	  particularly when using multiple segments), in favour of solely
	  using the timestamps/durations.

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2006-08-29  Edward Hervey  <edward@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	Don't rely on incoming buffers offset anymore, since it is completely
	broken when using multiple segments.
	Instead convert the incoming buffers timestamp to running time, and
	then convert that value to the offsets.
	Also inform GstSegment of the last outputted stop position, which is
	needed if we received several segments with an unknown stop value.

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2006-08-29  Thomas Vander Stichele  <thomas at apestaart dot org>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  fix buffer unreffing on a header push failure

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2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
	(gst_audio_rate_chain):
	Make the metadata of the buffer writable before changing its
	flags.

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2006-08-28  Wim Taymans  <wim@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	(gst_audio_rate_setcaps), (gst_audio_rate_init),
	(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
	(gst_audio_rate_chain), (gst_audio_rate_change_state):
	Fix audiorate some more.
	Reset and resync counters on flush and READY.
	Handle the DISCONT flag correctly.
	Use GstSegment to track position.
	Fail when not negotiated.
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	Fixes #353234.
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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Fix spelling.
	  Remove accidently included debug line.

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

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	Small cleanups.
	If a buffer is received with no caps, make the buffer metadata
	writable and set the caps, making sure that we don't screw up the
	refcounts.

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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
	(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
	  Fix memory leaks and misleading debug messages, add a couple of
	  comments.

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
	(gst_multi_fd_sink_render):
	  Do not use gst_buffer_make_writable() in a basesink render method,
	  as it may incorrectly unref the buffer. Instead, use convoluted
	  dance to avoid copying the buffer except when we need to.

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2006-08-25  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c:
	(gst_vorbis_enc_buffer_check_discontinuous):
	  Allow very small discontinuities in the timestamps. These we can't
	  do anything useful with anyway (because vorbis's timestamps have
	  only sample granularity), and are commonly produced by elements with
	  minor bugs. Allow up to 1/2 a sample out.
	  Fixes #351742.

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

	* tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
	(play_scrub_toggle_cb), (main):
	Add a checkbox to enable play scrubbing. Makes it possible to disable
	normal scrubbing.

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

	* tests/check/elements/.cvsignore:
	  make buildbot happy

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
	(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
	(gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
	(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
	(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	(gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
	(gst_ogm_text_parse_strip_trailing_zeroes),
	(gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
	(gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
	  Refactor ogm parse, do better input checking, misc. clean-ups.
	  Cache incoming events and push them once the source pad has
	  been created. Don't pass unterminated strings to sscanf().
	  Strip trailing zeroes from subtitle text output, since they
	  are not valid UTF-8. Don't push vorbiscomment packets on
	  the subtitle text pad. Output perfect streams if possible.

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2006-08-23  Wim Taymans  <wim@fluendo.com>

	* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	Waits for tasks to settle down so that we clean up correctly for 
	valgrind.

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
	  Unit test fixes: \377 is more likely to fit into 8 bits than \777;
	  actually return return value in taglists_are_equal.

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2006-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Fix crash due to broken bitstream parsing on x86-64: can't make
	  any assumptions about sizeof(struct) due to alignment/packing
	  differences on different architectures. Fixes #351790.

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

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
	(gst_riff_parse_chunk), (gst_riff_parse_file_header),
	(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
	(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
	(gst_riff_parse_info):
	Protect public functions against bad input.
	Do some cleanups.
	Fix documentation.

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

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Add voxware audio IDs (even if we can't play it) (#351795).

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

	* gst-libs/gst/riff/riff-media.c:
	(gst_riff_create_video_template_caps),
	(gst_riff_create_audio_template_caps),
	(gst_riff_create_iavs_template_caps):
	  Const-ify some arrays and use G_N_ELEMENTS instead
	  of wasting oodles of RAM on terminator bits.

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

	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_to_vorbiscomment_buffer):
	* tests/check/libs/tag.c: (GST_START_TEST):
	  And the same for _to_vorbiscomment_buffer(): allow
	  id_data_len == 0 for speex.

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2006-08-21  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-gdp.xml:
	* gst/gdp/Makefile.am:
	* tests/check/Makefile.am:
	  Move GDP plugin to -base from -bad.  Closes #347783.

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2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_from_vorbiscomment_buffer):
	  Allow id_data_len == 0 (needed for vorbis comments in Speex files).
	  Also add some checks to make sure we don't memcmp() beyond the end of
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	  vorbiscomment buffer if the ID to check for is larger than the buffer.
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	* tests/check/libs/tag.c: (GST_START_TEST):
	  Some more tests for gst_tag_list_from_vorbiscomment_buffer().

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2006-08-21  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
	(gst_vorbis_enc_set_metadata):
	  Use vorbis comment utility functions from libgsttag
	  instead of re-inventing the wheel (partially fixes #347091).

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

	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	Fix leaks. Wait for state transitions that might happen ASYNC, as well
	as some that won't.

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

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	Don't try to GObject scan the netbuffer as it's not a GObject.
	Fixes #351308.

	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	* gst-libs/gst/netbuffer/gstnetbuffer.h:
	Document GstNetBuffer.

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

	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Add testcase for caps-size-explosion

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

	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_get_unit_size), (set_structure_widths):
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	  Lower debug, use g_assert in _get_unit_size
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	* gst/audioresample/gstaudioresample.c:
	(audioresample_get_unit_size):
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	(gst_ffmpegcsp_get_unit_size):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
	  use g_assert in _get_unit_size

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
	(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
	(gst_rtp_buffer_get_payload_buffer):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Document GstRTPBuffer.
	Added function to efficiently strip payload headers.
	API: gst_rtp_buffer_get_payload_subbuffer()

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2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
	(gst_tag_to_vorbis_comments):
	  Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
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	  tags and deserialise them properly as well (#347091).
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	  Add some more gtk-doc blurbs and also some g_return_if_fail().

	* tests/check/libs/tag.c: (GST_START_TEST),
	(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
	  More tests.

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

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogg.c: (plugin_init):
	* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
	(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
	(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
	(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
	(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
	(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
	Added ogg-in-avi parser element. Fixes #140139.

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
	Fixed a bug in oggdemux debug code.

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	Recognise Ogg in the AVI extensible wave format.

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2006-08-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  Make buffer durations add up (duration should be next_ts-ts for
	  perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
	  from CVS.

	* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
	(test_buffer_timestamps), (cddabasesrc_suite):
	  Add unit test for the above.

	* tests/check/Makefile.am:
	  Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
	  to see what happens.

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

	* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
	(gst_alsasink_open):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
	(gst_alsasrc_open):
	Avoid setting and using a NULL device name.
	Print more info when we fail to open a device.

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2006-08-16  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_parse_extended_comment):
	  API: add gst_tag_parse_extended_comment() (#351426).

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
	  Add unit test for gst_tag_parse_extended_comment().

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

	* sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
	  Fix leak (#351502).

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

	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/gst-plugins-base-plugins.args:
	* gst/playback/gstplaybin.c:
	  Document playbin.
	  
	* 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-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.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 to CVS version.

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2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_set_property), (gst_play_bin_get_property),
	(value_list_append_structure_list),
	(gst_play_bin_handle_redirect_message),
	(gst_play_bin_handle_message):
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	  API: GstPlayBin::connection-speed
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	  Add "connection-speed" property; re-order redirect messages with
	  multiple redirect locations depending on the minimum bitrate if
	  that information is available and a connection speed is set
	  (#350399).

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2006-08-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Update max volume to the same value that the volume element uses.

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2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
	Less uglyness..

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2006-08-14  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	(gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
	Add some more debug info.
	Don't crash when a seek failed.
	Actually return the result of the seek instead of TRUE.
	Ignore multiple BOS pages with the same serial so that we don't create
	the same stream multiple times.
	Post an error when we fail to do the initial seek.

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

	* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
	(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
	Small code cleanup.

	* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
	(gst_alsa_mixer_new):
	Remove hack that always set the device to hw:0*.
	Properly find the card name for whatever device was configured.
	Do some better debugging.
	Fixes #350784.

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_set_property),
	(gst_alsa_mixer_element_change_state):
	Cleanups.
	Handle setting of a NULL device name better.

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2006-08-11  Wim Taymans  <wim@fluendo.com>

	* gst/adder/gstadder.c:
	Don't clip float values. Fixes #350900.

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2006-08-11  Andy Wingo  <wingo@pobox.com>

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	* gst/tcp/gsttcp.c: Really fix the build?

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	* gst/tcp/gsttcp.h: For now, always disable deprecation here --
	fixes the build.

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2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
	  Float caps shouldn't have a "signed" field.

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2006-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
	  Implement SEEKING query in its most basic form, so that we can
	  at least check if we're seekable or not (#350655).

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2006-08-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  The checks here are not even close to anything that would
	  justify MAXIMUM probability, lowering to POSSIBLE until someone
	  fixes the checks (case at hand: quicktime redirection files
	  might start with 00 00 01 XX and pass the checks here just
	  fine, see #350399).

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

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
	  Better detection for multipart/x-mixed-replace: accept leading
	  whitespaces before the boundary marker as well (as our very own
	  multipartmux used to produce) (#349068).

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2006-08-07  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	(gst_riff_create_audio_template_caps):
	  Detect DTS audio streams (#350157).

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2006-08-05  Andy Wingo  <wingo@pobox.com>

	* ext/theora/gsttheoraparse.h:
	* ext/theora/theoraparse.c (gst_theora_parse_class_init)
	(theora_parse_dispose, theora_parse_set_property)
	(theora_parse_get_property, theora_parse_munge_granulepos)
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	(theora_parse_push_buffer, theora_parse_change_state):
	API: GstTheoraParse::synchronization-points
	Add a property 'synchronization-points' to fix badly synchronized oggs.
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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/audio.c: (structure_contains_channel_positions),
	(fixed_caps_have_channel_positions), (GST_START_TEST),
	(audio_suite), (main):
	  Add a few tests for the channel position stuff in libgstaudio.

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
	(gst_alsa_detect_channels):
	* ext/alsa/gstalsasink.c:
	  Add support for cards that (only) do more than 8 channels,
	  like the Delta 44 (#345188).

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions):
	* gst-libs/gst/audio/multichannel.h:
	  API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
	  unspecified channel position and cannot be combined with any
	  of the other audio channel positions; adjust position layout
	  checks accordingly (#345188).

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Recognise ancient RealAudio files (see #349779).

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2006-08-03  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jens Granseuer  <jensgr at gmx net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Add typefinder for Interplay's MVE format (#348973).

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2006-08-02  Wim Taymans  <wim@fluendo.com>

	Patch by: Marcel Moreaux <marcelm at luon dot net>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_add_to_queue):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Handle RTP sequence number rollover.
	Disable jitterbuffer by default.

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2006-07-28  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audioresample/gstaudioresample.c: (audioresample_stop),
	(audioresample_set_caps):
	Don't leak references to the incoming caps. Clean them up when
	stopping.

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	(gst_video_scale_finalize):
	Don't leak our temporary pixel buffer.

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
	(GST_START_TEST), (simple_launch_lines_suite):

	Fix leaks and re-enable the test for valgrind checking.

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

	Patch by: Sjoerd Simons  <sjoerd at luon net>

	* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
	(plugin_init):
	  Add typefind function for multipart/x-mixed-replace (#348916).

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

	* gst/adder/gstadder.c: (gst_adder_setcaps),
	(gst_adder_query_duration):
	Fix leak in duration query.
	Reflow some docs and notes.

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2006-07-28  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
	(vorbisenc_suite):
	  Enable Andy's extra vorbisenc test, now that it passes. Also fix one
	  aspect of it.

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2006-07-28  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
	(gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
	(gst_vorbis_enc_push_buffer),
	(gst_vorbis_enc_buffer_check_discontinuous),
	(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	* ext/vorbis/vorbisenc.h:
	  Handle discontinuities in the input vorbis stream correctly,
	  so that the output is properly timestamped (and has good granulepos
	  values). Needs some oggmux fixes too.

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2006-07-27  Wim Taymans  <wim@fluendo.com>

	patch by: Kai Vehmanen <kv2004 eca cx>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_change_state):
	Don't send multiple newsegments with different formats.
	Fixes #348677.

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2006-07-26  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
	Make seeking in ogg more accurate again by doing the more correct
	granuletime to stream time conversion.

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2006-07-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_new_client):
	  debug a little more understandably
	  do not use goto as a substitute for break, especially if
	  break is also being used

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

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
	* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Remove GLib-2.6 compatibility cruft.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	Don't try to align a sample to an unknown value.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
	When the audio clock is slaved to another clock, never try to align
	samples but trust the rate interpolation algorithm.

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2006-07-24  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	Don't try to calculate silence samples, base class does this much
	better now.

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_acquire):
	Calculate silence samples correctly.

	* gst-libs/gst/audio/gstringbuffer.h:
	Add _CAST macro.

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2006-07-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
	  Limit search for the first markup tag to the first few kB of
	  the file. If we don't find one there, it's highly unlikely that
	  this is an XML(-ish) file.

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	* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
	test to the one in vorbisenc. Also commented out.

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	* tests/check/pipelines/vorbisenc.c: 
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	(test_discontinuity): New test, commented out until Mike lands
	some elite vorbisenc patches.

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	* tests/check/pipelines/theoraenc.c: Port to bufferstraw.
	Bufferstraw was actually factored out of these tests. Now we share
	code yay.

	* configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
	for bufferstraw addition to gstcheck.

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2006-07-21  Wim Taymans  <wim@fluendo.com>

	* ext/theora/theoradec.c: (clip_buffer):
	Better clipping.

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2006-07-21  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
	(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
	(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
	Fix leak.
	Avoid type casting when we can.

	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
	Fix mem leak.

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2006-07-20  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsamixerelement.c:
	(gst_alsa_mixer_element_change_state):
	  Make state change fail if the specified device can't be opened
	  for some reason.

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2006-07-20  Wim Taymans  <wim@fluendo.com>

	* gst/playback/test.c: (gen_video_element), (gen_audio_element),
	(cb_newpad), (main):
	Example of a small audio/video player using decodebin.

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2006-07-20  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
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	  Add 'fact' chunk id
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2006-07-19  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_change_state):
	Don't assert when not negotiated but post a meaningfull 
	error message. Fixes #347918.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	Add comment about better default MTU size.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	Small cleanups, start docs.

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2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Martin Szulecki

	* sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
	  If "device-name" is requested and the device is not
	  open, try to temporarily open it to obtain this
	  information (#342494).

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2006-07-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstid3tag.c:
	  Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).

	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Some more random const-ifications.

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2006-07-18  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.c:
	(gst_riff_create_video_template_caps):
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	  Add more FOURCCs (sort list to make stuff easier to find),
	  add comment what those 16 bytes in struct _gst_riff_strh according to
	  one avi-dumper are
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2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/multichannel.c:
	(gst_audio_check_channel_positions),
	(gst_audio_fixate_channel_positions):
	  Const-ify two arrays.

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2006-07-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
	  Fix typo, so that alsasink also advertises 8 channels
	  if that's supported (tags: can, worms, open, alsa, ph34r).

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2006-07-17  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	(gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
	*sigh*, when is the compiler going to warn when the comments
	are out-of-sync with the code.. Refix case of busted theora
	headers with 0 granule pos.

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2006-07-14  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_wait),
	(gst_base_rtp_depayload_change_state),
	(gst_base_rtp_depayload_set_property),
	(gst_base_rtp_depayload_get_property):
	Fix 99% cpu load by waiting for absolute times on the
	clock. Fixes #347300.

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2006-07-14  Andy Wingo  <wingo@pobox.com>

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	* ext/theora/gsttheoraparse.h: 
	* ext/theora/theoraparse.c (theora_parse_drain_event_queue)
	(theora_parse_push_headers, theora_parse_clear_queue)
	(theora_parse_drain_queue_prematurely, )
	(theora_parse_sink_event, theora_parse_change_state): Queue events
	until we initialized our state, like in vorbisparse.

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	* ext/vorbis/vorbisparse.h: 
	* ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
	(vorbis_parse_push_headers, vorbis_parse_clear_queue)
	(vorbis_parse_drain_queue_prematurely, )
	(vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
	until we have initialized our state. Fixes seeking after an
	initial pad block.

2006-07-14  Andy Wingo  <wingo@pobox.com>

Thomas Vander Stichele's avatar
Thomas Vander Stichele committed
1034
	Patch by: Iain Holmes <iaingnome@gmail.com>
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	* ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.

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2006-07-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Bump nano back to CVS

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=== release 0.10.9 ===

2006-07-13  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.9, "I walk the line"

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2006-07-14  Michael Smith  <msmith@fluendo.com>

	* tests/check/pipelines/vorbisenc.c: (stop_pipeline):
	  Move a g_cond_signal to earlier to avoid sometimes deadlocking
	  (commonly happens when running this test under valgrind) when trying
	  to remove the buffer probe.

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2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
	Fix missing g_unlock from the previous commit

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2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	(gst_xvimagesink_change_state):
	Implement a locking order to ensure we always take the object lock
	before the x_lock and never vice-versa.

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2006-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (find_compatibles):
	Fix a caps leak when linking (#347304)

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
	(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
	Don't leak shared memory resources. Use the object lock to protect
	against the xcontext disappearing while returning a buffer from the
	pipeline. (#347304)

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
	(vorbis_handle_comment_packet):
	gst_tag_list_merge() returns a new object. Take that into account when
	using it. This avoids memleak.
	Revert previous commit which is not needed.

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
	Reset the decoder in finalize so that all fields get cleared.

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

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_set_clock),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
	Don't try to post an error message when setting the clock fails
	as this can happen when adding an element to a bin which will then
	deadlock. Fixes #347296.

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	(vorbis_handle_type_packet):
	Post tag messages on the bus even if we're not initialized.
	If we're not initialized, we still postpone the event pushing of tags.

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

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Revert last two changes that broke the freeze.

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

	* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	basesink calculates silence sample correctly for us.

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

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	Calculate correct silence samples so we don't fill our ringbuffer
	with noise.

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2006-07-12  Edward Hervey  <edward@fluendo.com>

	* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	(gst_vorbis_dec_reset), (vorbis_dec_sink_event),
	(vorbis_handle_comment_packet), (vorbis_handle_type_packet):
	* ext/vorbis/vorbisdec.h:
	Delay sending events (newsegment, tags) until the decoder is properly
	initialized.
	Fixes #347295

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2006-07-11  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/audioconvert.c: (get_float_mc_caps),
	(get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
	  Patch from #347221 adding a test for audioconvert
	  channel remappings.

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

	* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
	(gst_ssa_parse_parse_line):
	  Don't include the terminating NUL in the buffer size,
	  it's only there for extra paranoia (would add random
	  '*' characters at the end of each subtitle since the
	  terminator itself is not valid UTF-8 technically).
	  Also fix indenting after boilerplate macro.

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

	* gst/playback/gstdecodebin.c: (close_pad_link):
	  Also emit 'unknown-type' signal (which should really be
	  called unhandled-type) if we found potential decoders/demuxers
	  in the registry but none of them worked in the end (as in the
	  case where the plugins don't exist any longer but are still
	  listed in the registry). Fixes #329798.

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2006-07-08  Andy Wingo  <wingo@pobox.com>

	* theoraparse.c (theora_parse_push_buffer)
	(theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
	Add some more debugging. Fix granulepos reconstruction in the face
	of discontinuities.

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2006-07-06  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init),
	(gst_base_audio_sink_provide_clock):
	Use gobject_class instead of G_OBJECT_CLASS (klass)

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_init),
	(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
	(gst_base_audio_src_get_time),
	(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
	(gst_base_audio_src_create_ringbuffer):
	Fix latency and buffer-time constants and properties ala basesink.
	Implement pull based scheduling. Fixes #346527.
	Set default blocksize in GstBaseSrc to 0, we default to pushing out
	one segment.
	Refuse slaving to another clock instead of silently not working.
	Only provide a clock when we are actually able to do so.
	Various small cleanups and compiler hints.

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2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Lutz Mueller <lutz at topfrose de>

	* gst/typefind/gsttypefindfunctions.c: (html_type_find),
	(plugin_init):
	  Add typefinding for text/html (#346581).

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2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	(xml_check_first_element), (xml_type_find), (smil_type_find):
	  Fix SMIL typefinding, make xml_check_first_element() more
	  useful.

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2006-07-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	(gst_play_base_bin_finalize), (decodebin_element_added_cb),
	(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
	* gst/playback/gstplaybasebin.h:
	  Protect list of elements with a subtitle-encoding property and
	  the subtitle encoding member itself with a lock of their own
	  instead of using the object lock. This prevents a dead-lock in
	  the element-remove callback in some circumstances when shutting
	  down playbin.

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

	* win32/common/libgsttag.def:
	Export some new functions.
	* win32/vs6/libgstogg.dsp:
	Add a link to libgsttag-0.10.lib.

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

	* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Some const-ification.

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2006-07-04  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
	Improve checking if we are dealing with a stream. Added some
	more uris that need buffering.

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

	* ext/vorbis/vorbisdec.c: (vorbis_do_clip):
	Remove unused variable.

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2006-07-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	  include lcov.mak
	* configure.ac:
	  add GCOV_LIBS to GST_LIBS

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2006-07-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Michael Sheldon  <webmaster at mikeasoft com>

	* ext/alsa/gstalsasrc.c:
	  Add 32 bps to template caps and increase channels range
	  from [1,2] to [1,MAX]. See #346326.

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

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Recognise 'WMVA' video codec fourcc (#345879).
	  
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2006-06-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	 
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c: 	 
	  Fixed nasty memory leak

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2006-06-26  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
	(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
	  fix logging

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2006-06-23  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
	(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
	Protect remove_fakesink using a mutex, so that we don't try and
	remove the fakesink simultaneously from multiple threads.

	When going from READY to PAUSED, restore the fakesink, so that
	it is there when decodebin gets reused.

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2006-06-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gsttcpclientsink.c:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversink.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/videorate/gstvideorate.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* sys/v4l/gstv4ljpegsrc.c:
	* sys/v4l/gstv4lmjpegsink.c:
	* sys/v4l/gstv4lsrc.c:
	* tests/examples/seek/scrubby.c:
	* tests/examples/seek/seek.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).

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

	* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
	  Second field in GEnumValue shouldn't be a description,
	  but a stringified version of the enum value.

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2006-06-22  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
	(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
	Avoid type checking in buffer casts.
	Avoid caps copy in buffer_alloc when we can.
	Use pad_peer_accept.

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

	* gst-libs/gst/tag/tag.h:
	  Oops, make that 'Since: 0.10.9'.

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2006-06-22  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: (register_tag_image_type_enum),
	(gst_tag_image_type_get_type):
	  API: add GstTagImageType enum to describe images contained
	  in image tags (#345641).

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

	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	  Fix warnings with gst-inspect: "buffers-min" property
	  should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
	  typo in property description.

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

	Patch by: Cody Russell <bratsche at gnome org>

	* gst/audioresample/gstaudioresample.c:
	(gst_audioresample_class_init):
	* gst/playback/gststreamselector.c:
	(gst_stream_selector_class_init):
	* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
	* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
	* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
	* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
	* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
	* gst/videotestsrc/gstvideotestsrc.c:
	(gst_video_test_src_class_init):
	* gst/volume/gstvolume.c: (gst_volume_class_init):
	  Avoid unnecessary class cast check in class_init
	  functions (#337747).

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2006-06-21  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
	(gst_text_overlay_video_chain):
	  g_markup_escape_text() REALLY doesn't like non-UTF8 input
	  and doesn't validate its input either (and neither did
	  textoverlay it seems). Let's do that then and fix #345206.

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2006-06-19  Wim Taymans  <wim@fluendo.com>

	* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	(gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
	(gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
	(find_syncframe), (find_limits), (assign_value),
	(count_burst_unit), (gst_multi_fd_sink_new_client),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
	(gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
	(gst_multi_fd_sink_change_state):
	* gst/tcp/gstmultifdsink.h:
	Added shiny new burst-on-connect methods.
	Add properties to control the minimal amount of data queued.
	Small cleanups.
	API: bytes-min property
	API: time-min property
	API: buffers-min property
	API: burst-unit property
	API: burst-value property
	API: add-full signal

	* gst/tcp/gsttcp-marshal.list:
	Added new marshaller code for the new signal.

	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(multifdsink_suite):
	Added testcases for new burst methods.

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2006-06-19  Edward Hervey  <edward@fluendo.com>

	* ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
	Implement clipping for accurate seeking.
	Closes #345225

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2006-06-19  Wim Taymans  <wim@fluendo.com>

	Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>

	* gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
	(gst_video_scale_transform):
	Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131

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

	* configure.ac:
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
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	  Fix --disable-external (can't set conditionals conditionally,
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	  #343602).

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/audioresample.c: (test_reuse),
	(audioresample_suite):
	  Add test case for bug #342789 fixed below.

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioresample/gstaudioresample.c:
	(gst_audioresample_class_init), (gst_audioresample_init),
	(audioresample_start), (audioresample_stop),
	(gst_audioresample_set_property), (gst_audioresample_get_property):
	  Implement GstBaseTransform::start and ::stop so that audioresample
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
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	  can clear its internal state properly and be reused instead of
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	  causing non-negotiated errors with playbin under some circumstances
	  (#342789).

	* tests/check/elements/audioresample.c: (setup_audioresample),
	(cleanup_audioresample):
	  Need to set element state here so that ::start and ::stop are
	  called.

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2006-06-16  Wim Taymans  <wim@fluendo.com>

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

	* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
	Parse extra data better, apparently it's right behind
	the normal strf header size. Fixes #343500.

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2006-06-16  Wim Taymans  <wim@fluendo.com>

	* ext/alsa/gstalsasink.c: (set_hwparams):
	If we fail to set the buffer_time and period_time alsa
	parameters, post a warning and leave alsa select a 
	default instead of failing. Fixes #342085

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/cdda/gstcddabasesrc.h:
	  Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
	  out in the header file and shouldn't be listed in the docs.

	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
1472
	  Must dereference pointer to fourcc in the debug statement.
1473

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2006-06-16  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	add remaining symbols into correct setions
	
	* gst-libs/gst/audio/gstringbuffer.c:
	fix incomplete docs
	
	* gst-libs/gst/audio/gstringbuffer.h:
	comment out not yet implemented function
	
	
	* gst-libs/gst/floatcast/floatcast.h:
	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	add short descriptions
	
	
	* gst-libs/gst/interfaces/propertyprobe.c:
	fix return value docs	
	
	* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	simplify debug logging
	
	* gst-libs/gst/riff/riff-read.h:
	sync function prototype and docs
	
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	remove left over symbol

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2006-06-16  Tim-Philipp Müller  <tim at centricular dot net>

	* autogen.sh:
	* configure.ac:
	* docs/Makefile.am:
	  Use GST_PLUGIN_DOCS macro in configure.ac, add
	  --enable-plugin-docs default to autogen.sh and use
	  ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).

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2006-06-15  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
	(gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
	(gst_ogg_demux_loop):
	Combine GstFlowReturn from the source pads to give a
	meaningfull result to the upstream peer or to stop the
	processing task in case of errors.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c: (cb_probe):
	  Try GST_TAG_CODEC as fallback when extracting the
	  codec name; more debug info.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Extract language tags from ogm subtitle streams, so that
	  the subtitle menu choices are labelled correctly in
	  Totem (fixes #344708).

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2006-06-14  Wim Taymans  <wim@fluendo.com>

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

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
	(gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
	(gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
	(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
	Fix various leaks. Fixes #343699.
	Add x-smoke mime type.

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2006-06-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-ids.h:
	  Add IDs for 'bext' chunks (see #343837).

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2006-06-12  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/subparse/samiparse.c: (sami_context_pop_state),
	(handle_start_font), (end_sami_element):
	  Honour font face tags in SAMI subtitles (#344503).

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* po/POTFILES.in:
	  add missing files containing translatable strings

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/tmpl/.cvsignore:
	  we don't want those *.sgml files in CVS either

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Stefan Kost committed
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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/.cvsignore:
	* tests/check/elements/.cvsignore:
	* tests/check/libs/.cvsignore:
	  ignore more

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	  also commiting the changed Makefile.am (added more libs to the
	  doc-build)

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2006-06-11  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* docs/libs/gst-plugins-base-libs.types:
	  first batch of reordering things, add index & hierarchy

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2006-06-11  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  use GST_PKG_CHECK_MODULES, cleans up output

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2006-06-10  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
	  Add support for burn:// URIs (#343385); const-ify things a bit,
	  use G_N_ELEMENTS instead of hard-coded array size.

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2006-06-10  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
	  Fix up broken entities before passing them to libxml *sigh*.
	  (#343303).
	  
Thomas Vander Stichele's avatar
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2006-06-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	  back to TRUNK

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Thomas Vander Stichele committed
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=== release 0.10.8 ===

2006-06-09  Thomas Vander Stichele <thomas at apestaart dot org>

	* configure.ac:
	  releasing 0.10.8, "Moar gij ziet mij nie"

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Thomas Vander Stichele committed
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2006-06-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.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:
	* win32/common/config.h:
	  0.10.7.2 prerelease

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2006-06-07  Thomas Vander Stichele  <thomas at apestaart dot org>

	* docs/libs/tmpl/gstaudio.sgml:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* docs/libs/tmpl/gsttuner.sgml:
	* docs/libs/tmpl/gstxoverlay.sgml:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/gstringbuffer.c:
	* gst-libs/gst/interfaces/colorbalance.c:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/interfaces/tuner.c:
	* gst-libs/gst/interfaces/xoverlay.c:
	  move last template doc snippets to source code and delete them

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2006-06-06  Michael Smith  <msmith@fluendo.com>

	* ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
	(theora_parse_drain_queue):
	  Mark DELTA_UNIT on non-keyframes.

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2006-06-03  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
	(gst_ring_buffer_samples_done):
	* gst-libs/gst/audio/gstringbuffer.h:
	Document better the fact that latency_time and buffer_time are values
	stored in microseconds, and not the usual GStreamer nanoseconds.
	Change the variables (compatibly) that store them from GstClockTime 
	to guint64 to make it more clear that they're not storing clock times.
	Also, remove the bogus property description that says the user can
	specify -1 to get the default value, since that's never been the case.

	When computing the default segment size for the ring buffer, make it
	an integer number of samples.

	When the sub-class indicates a delay greater than the number of
	samples we've written return 0 from the audio sink get_time method.

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* tests/check/elements/audioconvert.c: (set_channel_positions),
	(get_float_mc_caps), (get_int_mc_caps):
	* tests/check/elements/audioresample.c:
	* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
	* tests/check/elements/videorate.c:
	* tests/check/elements/videotestsrc.c: (GST_START_TEST):
	* tests/check/elements/volume.c:
	* tests/check/elements/vorbisdec.c:
	* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
	  Don't busy-wait in tests; this was causing test timeouts very
	  frequently when running under valgrind.

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2006-06-02  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/tcp/README:
	* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
	(gst_multi_fd_sink_remove_client_link),
	(gst_multi_fd_sink_client_queue_caps),
	(gst_multi_fd_sink_client_queue_buffer),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_render):
	* gst/tcp/gstmultifdsink.h:
	  make multifdsink properly deal with streamheader:
	  - streamheader is taken from caps
	  - buffers marked with IN_CAPS are not sent
	  - streamheaders are sent, on connection, from the caps of the
	    buffer where the client gets positioned to
	  - further streamheader changes are done every time the client
	    will receive a buffer with different caps
	* tests/check/elements/multifdsink.c: (GST_START_TEST),
	(gst_multifdsink_create_streamheader):
	  add tests for this

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2006-06-02  Michael Smith  <msmith@fluendo.com>

	* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	  Reinstate limit on channel count. Vorbis does not define the meaning
	  of > 6 channels, so they're just independent channels. Gstreamer
	  currently has no mechanism to represent N independent channels.