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2007-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/Makefile.am:
	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
	  Use gst_tag_freeform_string_to_utf8() here, which also takes
	  into account any character sets specified by the user via
	  environment variables.

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2007-12-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/audioconvert/Makefile.am:
	Also link to libm.

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2007-12-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	No need for floating point operations here. avoids having to link
	against the math library too.

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2007-12-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/descriptions.c: (formats),
	  (format_info_get_desc):
	* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
	  (GST_START_TEST):
	  Add one or two missing formats.  Generate ADPCM description
	  dynamically depending on layout/format.

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	  Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.

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2007-12-08  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Robin Stocker <robin dot stocker at gmx dot ch>

	* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
	  Some .srt files start with chunk number 0 and not chunk number 1,
	  recognise and accept those as well (fixes #502497).

	* tests/check/elements/subparse.c: (srt_input), (srt_input0),
	  (test_src):
	  Add unit test for the above.

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

	* gst/playback/gstplay-enum.c:
	(register_gst_autoplug_select_result),
	(gst_autoplug_select_result_get_type):
	* gst/playback/gstplay-enum.h:
	Add missing files.

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

	* gst/playback/Makefile.am:
	Group decodebin2 and uridecodebin into the same plugin so that they
	can share the GEnumType.

	* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
	(_gst_select_accumulator), (gst_decode_bin_class_init),
	(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
	(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
	(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
	Add signal to sort factories instead of the more awkward autoplug-select
	signal.
	Modify autoplug_select so that we can try, skip or expose the
	autopluggin of an element on a pad.

	* gst/playback/gstfactorylists.c: (compare_ranks),
	(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
	(element_filter), (gst_factory_list_get_elements),
	(gst_factory_list_debug), (gst_factory_list_filter):
	* gst/playback/gstfactorylists.h:
	Simplify the API, allow getting elements based on mask.

	* gst/playback/gstplay-marshal.list:
	Add some more marshallers.

	* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
	(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
	(autoplug_select_cb), (activate_group):
	Add support for managing non-raw sinks by providing a custom element and
	sink list to decodebin2.
	Try to plug non-raw sinks when decodebin2 using autoplug-select of
	decodebin2.

	* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
	(gst_play_sink_set_mode), (gst_play_sink_request_pad):
	* gst/playback/gstplaysink.h:
	Add support for raw and non-raw sinks. 
	Add support to force sinks selected by playbin2.
	Don't plug raw converters for non-raw sinks.

	* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
	(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
	(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
	(plugin_init):
	Use right accumulators.
	Proxy new signal.

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2007-12-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	Use runnning time as the base time instead of the timestamp.
	Spotted by Saur on IRC.

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2007-12-03  Edward Hervey  <bilboed@bilboed.com>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	Add 'WVC1' codec mapping for Windows Media VC-1 video codec.

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2007-12-03  Wim Taymans  <wim.taymans@gmail.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
	(gst_ogg_demux_read_chain):
	If we find a new serial number but it does not contain a BOS page, make
	sure we initialize the chain to NULL because else we will try to scan it
	and crash. Fixes #500763

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2007-11-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/Makefile.am:
	* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
	(get_feature_array), (decoders_filter), (sinks_filter),
	(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
	(gst_factory_list_filter):
	* gst/playback/gstfactorylists.h:
	Refactor some common code to filter factories and check caps compat.

	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	(gst_decode_bin_init), (gst_decode_bin_dispose),
	(gst_decode_bin_autoplug_continue),
	(gst_decode_bin_autoplug_factories),
	(gst_decode_bin_autoplug_select), (analyze_new_pad),
	(find_compatibles):
	* gst/playback/gstplaybin.c:
	* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
	(gst_play_bin_init), (gst_play_bin_finalize),
	(autoplug_factories_cb), (activate_group):
	* gst/playback/gstqueue2.c:
	* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
	(proxy_autoplug_continue_signal),
	(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
	(proxy_drained_signal):
	Add some more debug info and use factor filtering code.

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2007-11-26  Stefan Kost  <ensonic@users.sf.net>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/volume/gstvolume.c:
	* gst/volume/gstvolume.h:
	  Add GAP-flag support.

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2007-11-24  Julien MOUTTE  <julien@moutte.net>

	* tests/examples/seek/seek.c: (main): Increase the range of the
	rate selector as I would like to test QOS behavior at higher
	forward and reverse playback speed like say 64x.

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2007-11-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	(audioresample_query), (audioresample_query_type),
	(gst_audioresample_set_property):
	Implement latency query.

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2007-11-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_drain):
	Our EOS time contains the base_time, _wait_eos() expects a running_time
	so we have to subtract the base_time again before calling the function.
	This fixes an EOS regression where the base_time was added twice and EOS
	took longer and longer in certain situations.
	Fixes #498767.

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2007-11-21  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	(gst_base_audio_sink_set_provide_clock),
	(gst_base_audio_sink_get_provide_clock),
	(gst_base_audio_sink_set_slave_method),
	(gst_base_audio_sink_get_slave_method),
	(gst_base_audio_sink_set_property),
	(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
	(gst_base_audio_sink_none_slaving),
	(gst_base_audio_sink_handle_slaving):
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	Expose methods for some object properties so that subclasses can more
	easily configure them.
	Added slave method none, that completely disables slaving to the
	internal clock.
	API: gst_base_audio_sink_set_provide_clock()
	API: gst_base_audio_sink_get_provide_clock()
	API: gst_base_audio_sink_set_slave_method()
	API: gst_base_audio_sink_get_slave_method()

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_set_provide_clock),
	(gst_base_audio_src_get_provide_clock),
	(gst_base_audio_src_set_property),
	(gst_base_audio_src_get_property), (gst_base_audio_src_create):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	Expose methods for some object properties so that subclasses can more
	easily configure them.
	API: gst_base_audio_src_set_provide_clock()
	API: gst_base_audio_src_get_provide_clock()

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2007-11-20  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Joe Peterson <lavajoe at gentoo dot org>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	Fix compilation on FreeBSD (Gentoo). Fixes #498228.

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2007-11-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/libs/fft.c: (GST_START_TEST):
	Fix scaling to really have dB instead of something else.

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2007-11-19  Julien MOUTTE  <julien@moutte.net>

	* tests/examples/seek/seek.c: (main): There's a nice macro to check
	GTK version, use it.

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2007-11-19  Julien MOUTTE  <julien@moutte.net>

	* tests/examples/seek/seek.c: (main): Try to support stable version
	of GTK.

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2007-11-17  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/README:
	* gst/playback/test7.c:
	  Fix the build + little README update.

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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
	Add playbin2 seek pipeline.

Wim Taymans's avatar
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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/Makefile.am:
	* gst/playback/gstplayback.c: (plugin_init):
	* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
	(eos_cb), (about_to_finish_cb), (main):
	Add playbin2.
	Added gapless playback example.

	* gst/playback/gstplaybasebin.c:
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
	* gst/playback/gstqueue2.c:
	* gst/playback/test.c:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	(pad_removed_cb):
	* gst/playback/gststreaminfo.h:
	Change email.

	* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
	(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
	(gst_play_bin_dispose), (gst_play_bin_set_uri),
	(gst_play_bin_set_suburi), (gst_play_bin_set_property),
	(gst_play_bin_get_property), (gst_play_bin_handle_message),
	(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
	(drained_cb), (unlink_group), (activate_group),
	(setup_next_source), (gst_play_bin_change_state),
	(gst_play_bin2_plugin_init):
	Added raw first version of playbin2. Does chained oggs and gapless
	playback fine. No support for raw sinks yet. No visualisations or
	subtitles yet.

	* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
	(gst_play_sink_class_init), (gst_play_sink_init),
	(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
	(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
	(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
	(gst_play_sink_set_property), (gst_play_sink_get_property),
	(post_missing_element_message), (free_chain), (add_chain),
	(activate_chain), (gen_video_chain), (gen_text_element),
	(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
	(gst_play_sink_set_mode), (gst_play_sink_request_pad),
	(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
	(gst_play_sink_send_event), (gst_play_sink_change_state):
	* gst/playback/gstplaysink.h:
	Added Element that abstracts the sinks and their pipelines for playbin2.

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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
	(gst_selector_pad_class_init), (gst_selector_pad_init),
	(gst_selector_pad_finalize), (gst_selector_pad_reset),
	(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
	(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
	(gst_selector_pad_chain), (gst_stream_selector_get_type),
	(gst_stream_selector_base_init), (gst_stream_selector_class_init),
	(gst_stream_selector_init), (gst_stream_selector_set_property),
	(gst_stream_selector_get_linked_pad),
	(gst_stream_selector_getcaps),
	(gst_stream_selector_is_active_sinkpad),
	(gst_stream_selector_activate_sinkpad),
	(gst_stream_selector_get_linked_pads),
	(gst_stream_selector_request_new_pad),
	(gst_stream_selector_release_pad):
	* gst/playback/gststreamselector.h:
	Improve streamselector, make it select and unselect the current pad more
	intelligently. 
	Subclass GstPad for the sinkpads of the selector.
	Handle segments more correctly.
	Fix caps negotiation.
	Implement release_pad.

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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	(gst_decode_group_check_if_drained), (source_pad_event_probe),
	(remove_fakesink):
	Add drained signal fired when decodebin finishes decoding the data.
	Remove deprecated STATE_DIRTY message.

	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
	(analyse_source), (proxy_drained_signal), (make_decoder),
	(source_new_pad), (value_list_append_structure_list),
	(handle_redirect_message), (handle_message):
	Proxy the new drained signal.
	Handle pad removed from decodebin.
	Handle redirect messages by sorting multiple redirections based on the
	connection speed.

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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
	Fix leaking headers. Fixes #496761.

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2007-11-16  Wim Taymans  <wim.taymans@gmail.com>

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

	* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
	(gst_ximagesink_change_state):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
	Don't leak the PAR on errors. Fixes #496731.

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2007-11-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
	  (gst_tag_from_id3_user_tag):
	  Add mapping for audio cd discid tags, so we can extract
	  them from tags as well (see #347848). Also compare identifiers
	  in ID3v2 TXXX frames in a case-insensitive way to increase
	  compatibility when reading tags (discid vs. DiscID vs. DiscId).

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2007-11-16  Jan Schmidt  <jan.schmidt@sun.com>

	* gst-plugins-base.doap:
	Oops, fix the release name.

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2007-11-16  Jan Schmidt  <jan.schmidt@sun.com>

	* gst-plugins-base.doap:
	Add 0.10.15 release

Jan Schmidt's avatar
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2007-11-16  Jan Schmidt  <jan.schmidt@sun.com>

	* configure.ac:
	Back to CVS

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=== release 0.10.15 ===

2007-11-15  Jan Schmidt <jan.schmidt@sun.com>

	* configure.ac:
	  releasing 0.10.15, "No need to argue"

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2007-11-15  Jan Schmidt  <jan.schmidt@sun.com>

	* win32/vs6/libgstfft.dsp:
	Convert line endings to DOS.

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2007-11-15  Sebastien Moutte <sebastien@moutte.net>

	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstfft.dsp:
	* win32/MANIFEST:
	Add a project file for fft plugin and remove socket
	based plugin which don't build from the workspace.
	* win32/vs6/libgstaudio.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgsttag.dsp:
	Convert line endings back to DOS.
	Fixes #496724

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2007-11-14  Jan Schmidt  <jan.schmidt@sun.com>

	* win32/vs6/libgstinterfaces.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Convert line endings back to DOS

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2007-11-14  Jan Schmidt  <jan.schmidt@sun.com>

	* gst-libs/gst/fft/kiss_fft_f32.h:
	* gst-libs/gst/fft/kiss_fft_f64.h:
	* gst-libs/gst/fft/kiss_fft_s16.h:
	* gst-libs/gst/fft/kiss_fft_s32.h:
	Don't include malloc.h which doesn't exist on Mac OSX.
	Instead, pull in glib.h and use g_malloc/g_free for 
	consistency. Fixes: #496548

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2007-11-09  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gstdecodebin2.c:
	  Dont leak ghostpad. Fixes #475451.

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2007-11-09  Wim Taymans  <wim.taymans@gmail.com>

	* docs/design/design-decodebin.txt:
	* gst/playback/gstdecodebin2.c: (analyze_new_pad):
	Update some more docs and comments.

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2007-11-06  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastien Moutte  <sebastien moutte net>

	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
	  (gst_rtcp_unix_to_ntp):
	* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
	  Fix some C99-isms and and a missing function that some versions of
	  MSVC don't like too much (#494346).

	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstaudio.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgsttag.dsp:
	  Update vs6 projects files (#494346).

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2007-11-06  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>

	* win32/common/libgstaudio.def:
	* win32/common/libgstcdda.def:
	* win32/common/libgstinterfaces.def:
	* win32/common/libgstnetbuffer.def:
	* win32/common/libgstpbutils.def:
	* win32/common/libgstrtp.def:
	* win32/common/libgstrtsp.def:
	* win32/common/libgsttag.def:
	* win32/common/libgstvideo.def:
	  More missing symbols to export (fixes #493986).

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2007-11-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/fft/gstfftf32.c:
	* gst-libs/gst/fft/gstfftf32.h:
	* gst-libs/gst/fft/gstfftf64.c:
	* gst-libs/gst/fft/gstfftf64.h:
	* gst-libs/gst/fft/gstffts16.c:
	* gst-libs/gst/fft/gstffts16.h:
	* gst-libs/gst/fft/gstffts32.c:
	* gst-libs/gst/fft/gstffts32.h:
	* tests/check/libs/fft.c: (GST_START_TEST):
	Remove the magnitude and phase calculation functions as these have
	very special use cases and can't even be used for the spectrum
	element. Also adjust the docs to mention some properties of the used
	FFT implemention, i.e. how the values are scaled. Fixes #492098.

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2007-11-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
	  (finish_source):
	  Avoid crash when there are external subtitles (fixes #491722).

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2007-11-03  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsasink.c: (gst_alsasink_open):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
	  'Could not open resource for writing' is not an acceptable
	  error message when we can't open the audio device (see #492334),
	  even less so when we're trying to open it to record something.

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2007-11-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>

	* win32/common/libgstrtp.def:
	  Add some more missing symbols (#492813).

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2007-11-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Thijs Vermeir <thijsvermeir@gmail.com>

	* tests/check/elements/audioconvert.c: (verify_convert):
	  Add check to make sure that the out caps have a channel layout
	  set on them where they should have one.

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

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

	* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
	* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
	  Include our own _stdint.h instead of sys/types.h, makes MingW happy
	  (#492306).

	* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
	  Use _pipe directly, GLib doesn't have a pipe() macro any longer
	  (it disappeared in GLib 2.14.0) (#492306).

	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	  Fix includes and LIBS for win32/Mingw (#492306).

	* tests/examples/dynamic/addstream.c (pause_play_stream):
	  Use more portable g_usleep() instead of sleep() (#492306).

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

	Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>

	* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	  (gst_ring_buffer_parse_caps):
	  Return NULL instead of an enum that happens to be 0, fixes warning
	  on MSVC (#492114).

	* gst-libs/gst/audio/gstringbuffer.h:
	  No trailing commas in enum list (for gcc-2.9x).

	* gst/videotestsrc/videotestsrc.c: (random_char):
	  Make information loss explicit instead of implicitly truncating to
	  eight bits via the return value.  Fixes runtime error on MSVC when
	  using the debug CRT (#492114).

	* win32/common/config.h.in:
	  Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).

	* win32/common/libgstinterfaces.def:
	* win32/common/libgstrtp.def:
	  Export a few more symbols (#492114).

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2007-11-01  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	  Readd the deprecation guards, but preserve compilability.

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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
	  (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
	  Preserve channel layout when fixating the number of channels in the
	  output caps, or make sure there's a suitable channel position layout
	  set on the caps if required. Fixes #430677.

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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/decodebin.c: (test_text_plain_streams):
	  Make sure the pipeline really operates in push mode as it should
	  in this case.

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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/audio.h:
	  Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
	  compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
	  (ie. normal cvs builds) will fail.

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2007-10-31  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/Makefile.am:
	* gst-libs/gst/audio/audio.c:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/interfaces/mixer.c:
	  tell gtk-doc about the deprecation guard. Apply more doc fixes.

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2007-10-31  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/audio.c: (init_value_to_channel_layout),
	  (test_channel_layout_value_intersect), (audio_suite):
	  Add simple unit test to make sure GstValue intersection
	  of channel layouts works the way I think it does.

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2007-10-30  Stefan Kost  <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/interfaces/mixer.h:
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	  Fix the docs according to what gtk-doc complained about.

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2007-10-30  Stefan Kost  <ensonic@users.sf.net>

	* tests/icles/stress-playbin.c:
	  Fix the build.

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2007-10-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
	* gst/playback/gstdecodebin2.c: (analyze_new_pad):
	  Post nice/more useful error message if we don't have a decoder for
	  the primary type.

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2007-10-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
	Be a bit more useful, unblock the pads after we fired the no-more-pads
	signal so that we can use the signal to inspect and connect all pads
	without having to keep extra state outside of decodebin.

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2007-10-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gsturidecodebin.c:
	(gst_uri_decode_bin_autoplug_continue),
	(gst_uri_decode_bin_class_init), (no_more_pads_full):
	Implement default signal handler so that we return TRUE when nothing is
	connected.

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2007-10-28  Sebastian Dröge  <slomo@circular-chaos.org>

633
	* gst-libs/gst/riff/riff-media.c:
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	(gst_riff_wavext_add_channel_layout),
	(gst_riff_wave_add_default_channel_layout),
	(gst_riff_wavext_get_default_channel_mask),
	(gst_riff_create_audio_caps):
	Use the ALSA channel layout as default for wav files without channel
	layout information. This fixes playback of chan-id.wav on 5.1 systems
	for example. Also refactor the channel layout setting a bit and add
	more default channel orders. Fixes #489010.

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2007-10-26  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
	  GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
	  -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
	  instead.

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2007-10-25  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
	(gst_decode_bin_set_subs_encoding),
	(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
	(gst_decode_bin_get_property), (analyze_new_pad):
	Move subtitle encoding property to decodebin2 so that it can set the
	property value on all elements that it autoplugs and that require it.
	Make caps refcounting more consistent in get/set.

	* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
	(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), (proxy_unknown_type_signal),
	(proxy_autoplug_continue_signal),
	(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
	(make_decoder):
	Proxy properties and relevant signals from the internal decodebin.
	Make properties MT safe.

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2007-10-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
	* gst-libs/gst/tag/tags.c:
	  Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
	  GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).

	* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
	  Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).

	* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
	  (gst_tag_to_vorbis_comments):
	  Map new SORTNAME tags (these tags aren't even semi-official, so I'm
	  just mapping everything I found in the wild) (#414539).

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2007-10-24  Wim Taymans  <wim.taymans@gmail.com>

	Inspired by patch of: René Stadler <mail at renestadler dot de>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	(gst_decode_bin_autoplug_continue),
	(gst_decode_bin_autoplug_factories),
	(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
	(find_compatibles):
	* gst/playback/gstplay-marshal.list:
	Remove the autoplug-sort signal and replace it with a binding friendly
	autoplug-select signal.
	Add an autoplug-factories signal that can be used to generate a list of
	factories to try to autoplug.
	Add the GstPad to the autoplugging signal args as it might be needed to
	make a good factory selection.
	Fix up the marshallers for this. Fixes #407282.

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2007-10-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gsttagdemux.c:
	  Don't abort with an assertion if we receive a seek event with
	  a start type of NONE (see launchpad bug #155878).

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

	* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
	(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
	(gst_ximagesink_change_state), (gst_ximagesink_reset):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
	(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
	(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
	Make sure that before we clean up the X resources, we shutdown and join
	the event thread.
	Also make sure the event thread does not shut down immediatly after
	startup because the running variable is not yet correctly set.
	Fixes #378770. 

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2007-10-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstdecodebin.c: (new_pad), (type_found):
	Make the window for a race in typefind and shutting down smaller until
	we figure out the right locking here. Avoids #485753 usually.

	* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
	Remove unneeded lock causing a race in typefind and shutting down.
	Fixes #485753.

	* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
	Also remove sinks when going to NULL because we might not complete the
	state change to PAUSED, causing the PAUSED->READY state change not to
	happen.

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2007-10-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
	Also explicitly release the ringbuffer when going to NULL because it
	is required in the setcaps function, before the state change to PAUSED
	completes.

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2007-10-16  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/stress-playbin.c:
	  Does what it says on the tin.

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2007-10-15  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
	Fix queue negotiation. See #486758.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-10-12  Jan Schmidt  <Jan.Schmidt@sun.com>
760

Jan Schmidt's avatar
Jan Schmidt committed
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	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
	(gst_xvimagesink_xwindow_new),
	(gst_xvimagesink_update_colorbalance),
	(gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):

	Fix handling of some of the X atoms. If the last parameter is True,
	XInternAtom won't create the atom if it doesn't exist, and therefore
	might return None. This causes X errors on Xv implementations that
	don't provide the colour balance attributes.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gstvorbistag.c:
	* tests/check/libs/tag.c:
	  Extract vorbis comment LICENSE tags correctly.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Jason Kivlighn  <jkivlighn gmail com>

	* gst-libs/gst/tag/gstid3tag.c:
	* tests/check/libs/tag.c:
	  Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).

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

	* gst-libs/gst/tag/gsttagdemux.c:
	  Don't error out when a buggy downstream element doesn't
	  handle the newsegment event we send properly (especially
	  not without posting a meaningful error message on the
	  bus). See bug #471370 and launchpad bug #136264.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_drain):
	Use new basesink method to make our EOS drain interruptable.

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2007-10-10  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst-libs/gst/rtp/gstrtppayloads.c:
	Fix silly search-replace oversight.

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2007-10-09  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
	(gst_basertppayload_set_outcaps):
	Fix caps memleak. Fixes #484989.


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

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
	Fix debug output.

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

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_create):
	Also handle the case where there is no clock set on the audio source,
	like in the unit tests.

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2007-10-08  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst-libs/gst/rtp/gstrtppayloads.c:
	Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
	to avoid compiler warnings

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

	* gst/playback/gstdecodebin.c: (type_found),
	(gst_decode_bin_change_state):
	* gst/playback/gstdecodebin2.c: (type_found),
	(gst_decode_bin_change_state):
	Don't disconnect the have_type signal because we never reconnect it
	later on. Instead keep a variable to see if we already detected a type.

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

	* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
	(type_found):
	Unlink the signal handler when we found the type, we're not going to do
	anything sensible with more type_found signals anyway.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/gsttagdemux.c:
	  Don't leak caps.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/tag/gsttagdemux.c:
	* gst-libs/gst/tag/gsttagdemux.h:
	  API: add GstTagDemux base class for simple tag demuxers.

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Add GstTagDemux to docs.

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2007-10-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	(gst_rtp_buffer_get_payload_subbuffer):
	Fix bug introduced with last commit which inverted the logic and
	caused all buffers to be dropped. Fixes #483620.
	Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.

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

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Replace g_return_if_val (as it could be disabled), with regular return
	  and warning.

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2007-10-03  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/simple-launch-lines.c:
	  Print message name and not just number.

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2007-10-02  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	When slaved to the clock, don't try to align a sample with the previous
	one when going to PLAYING again.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/snapshot/snapshot.c:
	  Fix the build.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/sdp/gstsdpmessage.h:
	Add RFC 3556 bandwidth modifiers.

Wim Taymans's avatar
Wim Taymans committed
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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtppayloads.c:
	Update documentation.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
	(gst_rtp_payload_info_for_name):
	* gst-libs/gst/rtp/gstrtppayloads.h:
	Added new file and header to deal with payload info.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
	(gst_rtp_buffer_default_clock_rate):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Payload specific stuff is move to new headers.
	Implement _default_clock rate using the new payload function.

	* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
	(gst_sdp_parse_line):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Add some more comments.

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2007-10-01  Wim Taymans  <wim.taymans@gmail.com>

	* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	(sdp_check_header), (sdp_type_find), (plugin_init):
	Add typefind function for application/sdp.
	Remove some old dirac typefind code that was ifdeffed out.

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2007-09-29  Sebastien Moutte <sebastien@moutte.net>

	* win32/common/libgstaudio.def:
	Add new exported functions.
	* win32/vs6/grammar.dsp:
	Add autogeneration and copy of some autegenerated files from win32/common
	for rtsp library.
	* win32/vs6/libgstaudioconvert.dsp:
	Add gstaudioquantize.c to the build.
	* win32/vs6/libgstinterfaces.dsp:
	Add videoorientation.c to the build.
	* win32/vs6/libgstriff.dsp:
	Add libgsttag to the link libraries list.
	* win32/vs6/libgstvolume.dsp:
	Add liboil to the link.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstrtsp.dsp:
	* win32/common/libgstrtsp.def:
	Add files to build libgstrtsp library.
	
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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/snapshot/snapshot.c: (main):
	Print error when pipeline failed to construct.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	* gst-libs/gst/tag/gstid3tag.c:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Add mappings for the new GST_TAG_COMPOSER for vorbis comments
	  and ID3v2 tags.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/floatcast/floatcast.h:
	  Don't include config.h in an installed public header, this
	  might break compilation of applications that don't have such
	  a header and doesn't necessarily do what it's supposed to do
	  anyway (ie. check for the lrint/lrintf defines) (#442065).
	  Add docs for the various macros and document how this header
	  has to be used (link against libm, etc.); add a few FIXMEs;
	  include math.h for non-c99 code path.  Based on patch by
	  Jan Schmidt.
	  
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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
	of duplicating these macros in configure.ac.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/sv.po:
	* po/uk.po:
	  Updated translations to 0.10.14

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jakub Bogusz <qboosh@pld-linux.org>

	* po/pl.po:
	  Added Polish translation.

999 1000 1001 1002 1003 1004 1005
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Ilkka Tuohela <hile@iki.fi>

	* po/fi.po:
	  Added Finnish translation.

1006 1007 1008 1009 1010 1011 1012
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jorge González González <aloriel@gmail.com>

	* po/es.po:
	  Added Spanish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mogens Jaeger <mogens@jaeger.tf>

	* po/da.po:
	  Added Danish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Funda Wang <fundawang@linux.net.cn>

	* po/zh_CN.po:
	  Added Chinese (simplified) translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Alexander Shopov <ash@contact.bg>

	* po/bg.po:
	  Added Bulgarian translation.

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2007-09-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstqueue2.c: (gst_queue_push_one):
	Fix compilation wrt printf arguments.

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2007-09-20  Wim Taymans  <wim.taymans@gmail.com>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/snapshot/.cvsignore:
	* tests/examples/snapshot/Makefile.am:
	* tests/examples/snapshot/snapshot.c: (main):
	Add simple snapshot example program using appsink.

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2007-09-20  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/generic/states.c:
	  Improved state change unit test.

Stefan Kost's avatar
Stefan Kost committed
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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/.cvsignore:
	* tests/check/.cvsignore:
	  Ignore registries in any format.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Only copy timestamp on outgoing packets if the depayloader did not set
	one.
	Also copy duration on outgoing packets.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
	(gst_basertppayload_set_outcaps):
	Fix compilation because of missing %d in printf.
	When fixating caps, fixate what we can and throw away all remaining
	unfixed caps, subclasses should do something smart if they need to.

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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* ext/gnomevfs/gstgnomevfssrc.c:
	  Improve debug logs a bit and be more verbose if things go wrong.

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2007-09-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	(gst_text_overlay_set_property):
	* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_render):
	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
	(gst_rtcp_unix_to_ntp):
	* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
	* gst/playback/gstqueue2.c:
	* tests/examples/seek/seek.c: (set_scale):
	Fix a bunch of compile warnings shown with Forte.

	* gst/audiorate/gstaudiorate.c:
	Always pull in config.h before including any system headers.

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2007-09-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/playback/gstqueue2.c: (update_buffering),
	(gst_queue_locked_flush), (gst_queue_locked_enqueue),
	(gst_queue_handle_sink_event), (gst_queue_chain),
	(gst_queue_push_one), (gst_queue_sink_activate_push),
	(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
	Also fix #476514 for queue2.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_push_full),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state):
	Remove code to deal with RTP to GST time conversion, we now just copy
	the GST timestamp we receive to the outgoing buffers.
	Handle segment and flushes correctly.

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	When we have no valid input timestamp, use the previous rtp timestamp on
	the outgoing RTP packet instead of the RTP base time.

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2007-09-15  David Schleef  <ds@schleef.org>

	* ext/alsa/gstalsa.c:
	* ext/alsa/gstalsadeviceprobe.c:
	* ext/alsa/gstalsamixer.c:
	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	  Change alsa alloca's to malloc to fix warnings on gcc-4.2.

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2007-09-15  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
	Add some debug info when negotiating caps.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	A buffer with an empty payload is also a valid buffer.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
	(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
	(gst_basertppayload_change_state):
	Make sure we start our RTP timestamp from the random base RTP
	timestamp even if the buffer timestamp starts from some random value.

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2007-09-14  Wim Taymans  <wim.taymans@gmail.com>

	* configure.ac:
	* tests/examples/Makefile.am:
	* tests/examples/dynamic/.cvsignore:
	* tests/examples/dynamic/Makefile.am:
	* tests/examples/dynamic/addstream.c: (create_stream),
	(pause_play_stream), (message_received), (eos_message_received),
	(perform_step), (main):
	Add simple exmple app to demonstrate starting and pausing live and
	non-live bins in a PLAYING pipeline.

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2007-09-14  Julien MOUTTE  <julien@moutte.net>

	* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
	typefind for QCP files (RFC #3625)

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2007-09-13  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_init):
	Disable pull mode scheduling, we're not ready for it yet and it subtly
	breaks a lot of things.

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2007-09-12  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/elements/libvisual.c:
	  Test all libvisual plugins, not just the first one; this reproduces
	  bug #450336 quite easily.  Looks like a problem with the 'jess'
	  visualisation.

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2007-09-12  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/libvisual.c:
	  Add basic libvisual test case in an attempt to reproduce bug #450336.
	  Doesn't reproduce that bug, but some other crasher instead (invalid
	  free), at least with make elements/libvisual.forever and the bumscope
	  plugin on x86-64/gutsy. Leaving test disabled for now.

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2007-09-11  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
	(read_body), (gst_rtsp_connection_receive):
	Make sure we can not cancel in the middle of receiving a message.
	Fixes #475731.

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2007-09-11  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Josep Torra Valles <josep@fluendo.com>

	* gst/playback/gstplaybasebin.c:
	  Increase upper limit for audio queue a bit; fixes preroll problem
	  with playbin and decodebin2 when playing a quicktime trailer with
	  multichannel audio via http (#464666).

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2007-09-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_init),
	(gst_base_audio_src_provide_clock),
	(gst_base_audio_src_set_property),
	(gst_base_audio_src_get_property), (gst_base_audio_src_create):
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	Allow othe clocks than the internal clock to be used for the pipeline.
	Add property to disable clock provide.
	API: GstBaseAudioSrc::provide-clock

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2007-09-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gstdecodebin2.c:
	  Don't leak request pads. Fixes #475395.

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2007-09-09  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	(gst_ximage_buffer_class_init):
	* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	(gst_xvimage_buffer_class_init):
	Correctly chain up finalize with the parent class to prevent
	memory leaks. Fixes #474880.

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2007-09-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/volume/gstvolume.c: (volume_choose_func):
	* tests/check/elements/volume.c: (GST_START_TEST):
	Revert the latest change: floating point samples are allowed to
	have any value, not only values in the range [-1,1]. Thanks to Andy
	Wingo for noticing.
	Also fix processing of int32 samples with volumes > 4 by making the
	unity value smaller which prevents overflows.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* tests/check/libs/rtp.c:
	  Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Fix up GstRTPHeader helper struct so that compilers will not under
	  any circumstances add padding in between our fields, as currently
          happens with MSVC on win32, because that would lead to us sending
	  out RTP payloads with broken RTP headers (#471194).
	  Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
	  
	* tests/check/Makefile.am:
	* tests/check/libs/.cvsignore:
	* tests/check/libs/rtp.c:
	  Add some simple unit tests for GstRTPBuffer. Some are disabled
	  because the code tested still needs fixing (set_csrc() does not work).

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	* win32/MANIFEST:
	* win32/common/gstrtsp-enumtypes.c:
	* win32/common/gstrtsp-enumtypes.h:
	* win32/common/interfaces-enumtypes.c:
	* win32/common/interfaces-enumtypes.h:
	* win32/common/multichannel-enumtypes.c:
	  Add rtsp enumtypes (#474384) and update others.

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2007-09-06  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Fix configure check for HAVE_LIBXML_HTML.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/.cvsignore:
	  Ignore more, in case the build bots work again one day.

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2007-09-06  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/fft/Makefile.am:
	* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
	* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
	* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
	* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
	* gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
	* gst-libs/gst/fft/gstfft.h:
	* gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
	(gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
	(gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
	* gst-libs/gst/fft/gstfftf32.h:
	* gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
	(gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
	(gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
	* gst-libs/gst/fft/gstfftf64.h:
	* gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
	(gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
	(gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
	* gst-libs/gst/fft/gstffts16.h:
	* gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
	(gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
	(gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
	* gst-libs/gst/fft/gstffts32.h:
	* gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
	(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	(kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
	(kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
	* gst-libs/gst/fft/kiss_fft_f32.h:
	* gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
	(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	(kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
	(kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
	* gst-libs/gst/fft/kiss_fft_f64.h:
	* gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
	(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	(kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
	(kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
	* gst-libs/gst/fft/kiss_fft_s16.h:
	* gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
	(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	(kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
	(kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
	* gst-libs/gst/fft/kiss_fft_s32.h:
	* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
	(kiss_fftr_f32), (kiss_fftri_f32):
	* gst-libs/gst/fft/kiss_fftr_f32.h:
	* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
	(kiss_fftr_f64), (kiss_fftri_f64):
	* gst-libs/gst/fft/kiss_fftr_f64.h:
	* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
	(kiss_fftr_s16), (kiss_fftri_s16):
	* gst-libs/gst/fft/kiss_fftr_s16.h:
	* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
	(kiss_fftr_s32), (kiss_fftri_s32):
	* gst-libs/gst/fft/kiss_fftr_s32.h:
	* gst-libs/gst/fft/kiss_version:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	Add libgstfft, a FFT library based on Kiss FFT which is
	BSD licensed. Supported sample formats are int16, int32,
	float and double. For those formats a real FFT and IFFT
	can be done, different windowing functions can be applied
	and functions for extracting the magnitude and phase exist.
	Fixes #468619.

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	Integrate libgstfft into the docs.

	* tests/check/Makefile.am:
	* tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
	Add unit tests for libgstfft, currently only testing the FFT.
	Unit tests for IFFT will follow soon.

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

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
	(gst_sdp_message_init), (gst_sdp_message_uninit),
	(is_multicast_address), (gst_sdp_message_as_text),
	(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
	(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
	(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
	(gst_sdp_media_init), (gst_sdp_media_uninit),
	(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
	(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
	(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
	(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
	(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Separate INIT_ARRAY() and related macros into two versions, one for
	structures and one for pointers (e.g., INIT_ARRAY() and
	INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
	lists of emails and phone numbers.
	Add missing const as appropriate.
	Change all gint to guint since they all actually represent unsigned
	values.
	Do not use time as a variable name as it shadows the global time().
	Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
	Actually implement gst_sdp_message_add_time().
	Make gst_sdp_message_add_time() take repeat times as an argument.
	Store repeat times in GstSDPTime as a GArray rather than as gchar**.
	Corrected the definition of gst_sdp_media_get_bandwidth() (was
	misspelled as badwidth).
	gst-indented and a little clean up. Fixes #471067.

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2007-09-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_process_double), (volume_process_double_clamp),
	(volume_process_float_clamp):
	Correctly clamp float/double samples in the [-1.0,1.0] range to
	prevent weird effects.
	* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	Add unit tests for all samples types that had none before.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  Need to include stdlib.h for abs() here too.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gststreaminfo.c:
	  Fix build.

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2007-09-05  Stefan Kost  <ensonic@users.sf.net>

	* gst/playback/gststreaminfo.c:
	  Clean up some half-disabled code and comment.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

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

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	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	(gst_base_rtp_payload_audio_handle_event):
	Return FALSE from the event handler to let the parent class handle the
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	event. Fixes #446766.
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	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
	Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.

	* gst-libs/gst/rtp/gstbasertppayload.c:
	Bump the MTU to 1400.

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2007-09-03  Johan Dahlin  <jdahlin@async.com.br>

	* gst/typefind/gsttypefindfunctions.c (plugin_init): 
	Add an audio/x-nsf typefind function for the nsfdec element.

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2007-09-03  Renato Filho  <renato.filho@indt.org.br>
	* gst/playback/gstplaybasebin.c:
	Included "myth://" on stream_uris list for enable buffering to mythtv files

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
	(gst_rtcp_unix_to_ntp):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix parsing of RB blocks.
	Fix docs.
	Added helper functions to convert to/from UNIX and NTP time.
	API: gst_rtcp_ntp_to_unix()
	API: gst_rtcp_unix_to_ntp()

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
	(gst_rtp_buffer_get_header_len),
	(gst_rtp_buffer_get_extension_data),
	(gst_rtp_buffer_get_payload_subbuffer),
	(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
	(gst_rtp_buffer_ext_timestamp):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Fix some more docs.
	Implement handling of packets with extensions.
	Fix padding check in _validate().
	Added function to get extension data.
	API: gst_rtp_buffer_get_header_len()
	API: gst_rtp_buffer_get_extension_data()

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

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_class_init),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Add some more docs for the queue-delay property and fix a typo in a
	comment.

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	Fix typo.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
	(gst_base_audio_sink_change_state):
	When skew slaving, try to hover around the middle of a segment so that
	we at most drift by half a segment.
	If we are aligning in the oposite direction of the clock skew, we don't
	have to resync.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_setcaps),
	(gst_base_rtp_depayload_set_gst_timestamp):
	Be less silly with the segment start, just apply the clock-base to the
	timestamp.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	(gst_base_rtp_depayload_class_init),
	(gst_base_rtp_depayload_finalize),
	(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	(gst_base_rtp_depayload_handle_sink_event),
	(gst_base_rtp_depayload_set_gst_timestamp),
	(gst_base_rtp_depayload_change_state):
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	Deprecate the queue handling thread thing and remove the code.
	Use new method to calculate the extended timestamp.

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2007-08-31  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_packet_sdes_copy_entry):
	Use g_strndup which does exactly what we want.

	* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
	(gst_rtp_buffer_ext_timestamp):
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Add helper function to compare seqnums.
	Add helper function to calculate extended timestamps.
	API: gst_rtp_buffer_compare_seqnum()
	API: gst_rtp_buffer_ext_timestamp()

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2007-08-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	(gst_rtcp_packet_sdes_get_entry),
	(gst_rtcp_packet_sdes_copy_entry):
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	Fix and document SDES item data function.
	Add new function that makes a proper copy of SDES item data.
	API: gst_rtcp_packet_sdes_copy_entry()

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* gst/Makefile.am:
	  The tcp and subparse plugins are under gst, but not totaly free of
	  dependencies. Handle selection inconfigure.ac, so that they show up
	  on the final list of what is build and what is not. Maybe they should
	  better be moved to ext.

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2007-08-30  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Daniel Díaz  <yosoy@danieldiaz.org>

	* configure.ac:
	* gst/Makefile.am:
	  Check if libxml provides HTML parser which subparse needs.
	  Fixes #451970.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/alsa/gstalsa.c:
	  Fix typo and compilation on big endian systems.

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2007-08-29  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstssaparse.c:
	  Convert SSA newline codes into actual newline characters (#470766).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* tests/check/libs/pbutils.c:
	  API: also add gst_install_plugins_supported() while we're at it
	  (see #470456).

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2007-08-28  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/missing-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.h:
	* tests/check/libs/pbutils.c:
	  API: add gst_missing_*_installer_detail_new() convenience API so
	  that applications that know exactly what they're missing can request
	  installer detail strings for those items directly instead of having
	  to first create a dummy missing-plugin message and then get the
	  installer detail string from that.  Fixes #470456.

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2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstdecodebin.c: (close_pad_link):
	We need to set up delayed-linking whenever the caps are non-fixed,
	not just when there are multiple types - use gst_pad_is_fixed()
	to test.

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2007-08-26  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_plugin_message_get_installer_detail):
	  Add missing separator in PID fallback case.

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2007-08-24  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/alsa/Makefile.am:
	There is no GST_PLUGINS_BASE_LIBS defined.
	
	* ext/alsa/gstalsa.c:
	* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
	Add support for ALSA 24-bit formats.
	snd_pcm_delay can return an error code, especially
	during XRUNS. In that case, the best we can do is assume
	delay = 0.

	* gst/audioconvert/Makefile.am:
	Add flags from -base before any more-remote dependencies.

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2007-08-23  Sebastian Dröge  <slomo@circular-chaos.org>

1632
	Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
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	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_set_volume),
	(gst_volume_init), (volume_process_int32),
	(volume_process_int32_clamp), (volume_process_int24),
	(volume_process_int24_clamp), (volume_process_int16),
	(volume_process_int16_clamp), (volume_process_int8),
	(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
	* gst/volume/gstvolume.h:
	Add support for int32, int24 and int8 to the volume element.
	Fixes #445529.

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2007-08-23  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/examples/Makefile.am:
	  Fix even more.

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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* ext/gnomevfs/gstgnomevfssrc.c:
	* ext/gnomevfs/gstgnomevfssrc.h:
	* gst-libs/gst/Makefile.am:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	* sys/v4l/v4lsrc_calls.c:
	* tests/examples/Makefile.am:
	* win32/common/config.h:
	  Revert unwanted commit. many thanks to moap. I want a fix for 
	  https://thomas.apestaart.org/moap/trac/ticket/239

Stefan Kost's avatar
 
Stefan Kost committed
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2007-08-23  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  Move passthrough below gst_object_sync_values(). Fixes #442654.

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2007-08-22  Wim Taymans  <wim.taymans@gmail.com>

	* gst-libs/gst/audio/audio.c:
	Clarify the docs a little.

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2007-08-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/volume/gstvolume.c:
	  Enable liboil for float and add more details about problems with
	  int16.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
	Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	When calculating the first timestamp of the buffers, don't go below 0
	and clip the samples because the offset was on the eos page.
	Fixes #466717.

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2007-08-21  Wim Taymans  <wim.taymans@gmail.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
	(gst_ogg_demux_collect_chain_info):
	Also submit the eos page when trying to find the first timestamp.
	See #466717.

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

	* gst-libs/gst/audio/audio.h:
	Use gst_util_uint64_scale() instead of doing the math
	with double for GST_FRAMES_TO_CLOCK_TIME() and
	GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
	prevents rounding errors. Fixes #467667.

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

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
	(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	Small cleanups.
	On shutdown, don't read the control socket yet.
	Set timeout value correctly in all cases.
	Add function to check if the server accepts reads or writes.
	API: gst_rtsp_connection_poll()

	* gst-libs/gst/rtsp/gstrtspdefs.h:
	Fix compilation with -pedantic.
	Add enum for _poll.

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

	Patch by: Olivier Crete  <tester at tester ca>

	* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
	(gst_basertppayload_getcaps):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Add getcaps vfunc to basertppayload. See #465146.

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

	* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
	Only post buffering messages when we are a stream.

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2007-08-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.c:
	  Small docs fix and addition.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/icles/.cvsignore:
	* tests/icles/Makefile.am:
	* tests/icles/test-textoverlay.c:
	  Add a dumb little test for textoverlay alignments.

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2007-08-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams  <dcbw redhat com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  API: add "line-alignment" property (#459334). Add gtk-doc blurb for
	  "silent" property so there's a Since tag in the API reference.

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

	* gst-libs/gst/rtp/gstbasertppayload.c:
	(gst_basertppayload_set_outcaps):
	* gst-libs/gst/rtp/gstbasertppayload.h:
	Improve caps negotiation so that downstream elements can confiure
	certain RTP properties by fixing them on the caps. See #465146.
	Add docs.

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2007-08-11  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstbasertpdepayload.c:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Mark as deprecated some macros which were presumably meant to be
	  private API and accidentally exposed in the public header file.
	  Also actually _init() lock (only works at the moment because the
	  struct is zeroed out when created and the initial values in the
	  mutex struct are zeroes too). (#459585)

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

	* docs/libs/Makefile.am:
	  Remove cruft and do some cleanups.

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	  Prepare for comming gtkdoc features (rebase against online docs).

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2007-08-10  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Debug output fixes.
	* tests/check/elements/audiorate.c: (do_perfect_stream_test),
	(GST_START_TEST):
	  Change the number of buffers used; 500 is too many and leads to
	  timeouts.

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2007-08-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstqueue2.c:
	* gst/videorate/gstvideorate.c:
	  Printf format fixes (#465028).

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2007-08-09  Michael Smith <msmith@fluendo.com>

	* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If we have a large (> 1 second) discontinuity, push a series of
	  smaller buffers rather than a single very large buffer. Avoids
	  unreasonably large single buffer allocations when encountering a
	  large gap.
	* tests/check/elements/audiorate.c: (GST_START_TEST),
	(audiorate_suite):
	  Add a test for this.

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2007-08-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybasebin.c: (group_commit),
	(queue_remove_probe), (queue_threshold_reached):

	Patch by: Josep Torra Valles <josep@fluendo.com>
	Fixes: #465015
	Make sure we remove the check_queues buffer probe from the 
	correct queue to avoid racily going back to "buffering 99%" when
	buffering is actually complete.

	Also, fix the spelling of Josep's surname in the ChangeLog.

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2007-08-09  Stefan Kost  <ensonic@users.sf.net>

	* ext/ogg/gstoggmux.c:
	  Do not leak oggmux instance.
	
	* ext/vorbis/vorbisenc.c:
	  Also log values.

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

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

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

	patch by: Yang Hong <hongyang@redflag-linux.com>

	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextoverlay.h:
	  Add 'silent' property to GstTimeOverlay. Fixes #462979

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

1856
	Patch by: Josep Torra Valles <josep@fluendo.com>
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	* docs/plugins/gst-plugins-base-plugins.args:
	* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
	(gst_uri_decode_bin_get_property), (gen_source_element):
	Add connection-speed property. Fixes #464690.

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2007-08-07  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>

	* configure.ac:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect):
	Fix compilation on windows. Fixes #464320.

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2007-08-07  Wim Taymans  <wim.taymans@gmail.com>

1876
	Patch by: Josep Torra Valles <josep@fluendo.com>
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	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_init), (queue_threshold_reached),
	(gen_source_element), (setup_substreams),
	(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
	(gst_play_base_bin_get_streaminfo_value_array):
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	(gst_play_bin_set_property), (gst_play_bin_get_property),
	(gst_play_bin_handle_redirect_message):
	Move connection-speed property from playbin to playbasebin so that we
	can also configure it in source elements that have the connection-speed
	property. Fixes #464028.
	Add some debug info here and there.

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2007-08-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
	Properly respond to conversion queries. Fixes #464079.

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

	* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
	(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
	(gst_audio_test_src_init_sine_table),
	(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
	* gst/audiotestsrc/gstaudiotestsrc.h:
	Add float/double and int32 support to audiotestsrc. Fixes #460422.
	Also set the default volume to the default value specified in the
	GParamSpec.

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

	Patch by: Jens Granseuer <jensgr at gmx dot net>

	* gst/audioconvert/gstaudioquantize.c:
	Fix C89 incompatibilities and spelling of explanations. Fixes #463215.

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

	* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
	Add rdt manager for rdt transport.
	Fix parsing of RDT transport.

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

	* configure.ac:
	Back to CVS

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Jan Schmidt committed
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=== release 0.10.14 ===

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

	* configure.ac:
	  releasing 0.10.14, "Light Years Ahead"

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

	* tests/check/libs/audio.c: (GST_START_TEST):
	Fix the test to reflect the behaviour of gst_audio_clip_buffer.

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

	* gst-libs/gst/audio/audio.c:
	When clipping a buffer with no timestamp, assume it is
	within the segment without warnings.

	Fixes: #460978

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

	* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
	Fire the signal on the object, not the interface.

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

	* gst-libs/gst/rtsp/.cvsignore:
	Ber. Don't include the full path, idiot.

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

	* gst-libs/gst/rtsp/.cvsignore:
	Ignore generated files.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/interfaces-marshal.list:
	* gst-libs/gst/interfaces/rtspextension.c:
	* gst-libs/gst/interfaces/rtspextension.h:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtsp.h:
	* gst-libs/gst/rtsp/gstrtspextension.c:
	(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_detect_server),
	(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	(gst_rtsp_extension_configure_stream),
	(gst_rtsp_extension_get_transports),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/rtsp/gstrtspextension.h:
	* gst-libs/gst/rtsp/rtsp-marshal.list:
	Move the rtspextension.h interface into gstrtspextension.h
	as part of libgstrtsp instead of libgstinterfaces, because it's
	only for use within plugins, not applications. 
	Add stuff to do the enum & marshal generation needed in libgstrtsp now.
	Use the GST_TYPE_RTSP_RESULT enum type for the return value of the 
	signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
	is abstract.

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

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/interfaces-marshal.list:
	* gst-libs/gst/interfaces/rtspextension.c:
	(gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/interfaces/rtspextension.h:
	Fix marshaller for the send signal.
	Add URL to stream selection interface method.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst-libs/gst/riff/Makefile.am:
	Pull in our dependencies from -base before those from outside.

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	API: gst_rtsp_base64_decode_ip()
	Added function to decode Base64 in-place.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/libs/.cvsignore:
	Ignore the mixer test binary.

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2007-07-26  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	Gratuitous comment change to trigger a rebuild on the buildbots.

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

	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
	(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
	(gst_sdp_media_get_format), (gst_sdp_media_get_information),
	(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
	(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
	(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Constify args where we can.

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

	* gst-libs/gst/interfaces/Makefile.am:
	* gst-libs/gst/interfaces/rtspextension.c:
	(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	(gst_rtsp_extension_detect_server),
	(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	(gst_rtsp_extension_configure_stream),
	(gst_rtsp_extension_get_transports),
	(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	* gst-libs/gst/interfaces/rtspextension.h:
	Move interface for RTSP extensions from -good to here.
	Added helper methods to invoke interface methods.