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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Olivier Crete  <tester at tester ca>

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

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Zeeshan Ali  <zeenix gmail com>

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

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-20  Stefan Kost  <ensonic@users.sf.net>

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

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

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2007-04-19  Michael Smith  <msmith@fluendo.com>

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

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2007-04-18  Stefan Kost  <ensonic@users.sf.net>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-04-15  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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2007-04-14  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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2007-04-13  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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2007-04-13  Jan Schmidt  <thaytan@mad.scientist.com>

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

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

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

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

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

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

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

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

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2007-04-12  Michael Smith  <msmith@fluendo.com>

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

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

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

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

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

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

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

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2007-04-10  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Patch by: René Stadler  <mail at renestadler de>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	Add min-ptime property to RTP base audio payloader. Patch by
	olivier.crete@collabora.co.uk.
	Fixes #415001

	Indentation/whitespace/documentation fixes.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Patch by: Kamil Pawlowski  <kamilpe gmail com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-03-09  Michael Smith  <msmith@fluendo.com>

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

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2007-03-09  Thomas Vander Stichele  <thomas at apestaart dot org>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2007-03-07  Tim-Philipp Müller  <tim at centricular dot net>

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

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

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

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

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

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

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

Jan Schmidt's avatar
Jan Schmidt committed
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2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

Jan Schmidt's avatar
Jan Schmidt committed
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=== release 0.10.12 ===

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

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

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

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

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	Fix regression that made GStreamer skip the first samples of audio.
	Fixes #414684.

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

	* configure.ac:
	Bump version to 0.10.11.3 pre-release

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2007-03-05  Sebastian Dröge  <slomo@circular-chaos.org>

	* po/POTFILES.in:
	  Update paths for the rename from utils to pbutils to fix the build.

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2007-03-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/pbutils/Makefile.am:
	  Change directory to install headers in from gst/utils to gst/pbutils
	  as well.

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2007-03-05  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/interfaces/mixer.c:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/descriptions.c:
	(gst_pb_utils_get_source_description),
	(gst_pb_utils_get_sink_description),
	(gst_pb_utils_get_decoder_description),
	(gst_pb_utils_get_encoder_description),
	(gst_pb_utils_get_element_description),
	(gst_pb_utils_add_codec_description_to_tag_list),
	(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
	* gst-libs/gst/pbutils/descriptions.h:
	* gst-libs/gst/pbutils/install-plugins.c:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/pbutils/missing-plugins.c:
	(gst_missing_uri_source_message_new),
	(gst_missing_uri_sink_message_new),
	(gst_missing_element_message_new),
	(gst_missing_decoder_message_new),
	(gst_missing_encoder_message_new),
	(gst_missing_plugin_message_get_description):
	* gst-libs/gst/pbutils/missing-plugins.h:
	* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
	* gst-libs/gst/pbutils/pbutils.h:
	* gst-libs/gst/utils/Makefile.am:
	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/base-utils.h:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/descriptions.h:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/install-plugins.h:
	* gst-libs/gst/utils/missing-plugins.c:
	* gst-libs/gst/utils/missing-plugins.h:
	* gst-plugins-base.spec.in:
	* gst/playback/Makefile.am:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybasebin.c: (setup_subtitle),
	(gen_source_element):
	* gst/playback/gstplaybin.c: (plugin_init):
	* tests/check/Makefile.am:
	* tests/check/libs/pbutils.c: (GST_START_TEST),
	(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
	* tests/check/libs/utils.c:
	  rename utils to pbutils

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

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
	* docs/plugins/gst-plugins-base-plugins-sections.txt:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
	Add documentation for decodebin2 that indicates that the API
	is still unstable.

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

	* configure.ac:
	Update to 0.10.11.2 (0.10.12 pre-release)

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_async_play):
	base time is irrelevant here.

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

	* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	Improve debugging.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_query), (gst_base_audio_sink_event),
	(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
	Improve latency and clock slaving calculations.
	Improve slave clock calibration.

	* gst-libs/gst/audio/gstringbuffer.c:
	(gst_ring_buffer_commit_full):
	When we are asked to render N sample to 0 bytes, return N.

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

	* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	(gst_alsasink_write), (gst_alsasink_reset):
	* ext/alsa/gstalsasink.h:
	Remove unused dispose function.
	Rename lock to not interfere with alsasrc lock.

	* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
	(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
	(gst_alsasrc_read), (gst_alsasrc_reset):
	* ext/alsa/gstalsasrc.h:
	Implement finalize function.
	Use lock to protect alsa access.
	Implement _reset.
	Fine tune sw params.

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

	* configure.ac:
	  Convert to new AG_GST style.

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

	Patch by: Ed Catmur <ed at catmur dot co dot uk>

	* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
	(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
	Fix race condition when rapidly switching visualisations in playbin.
	Fixes #401029.

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2007-02-28  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Include local stuff before system installed things in LDFLAGS and
	CFLAGS.

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

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
	Improve debugging.

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

	* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
	(gst_v4lsrc_fixate), (gst_v4lsrc_query):
	* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
	Fix duration and timestamping, taking latency into account.
	Implement latency query.

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

	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	(gst_audio_clock_new):
	Fix clock name.

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_init), (gst_base_audio_sink_query):
	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
	(gst_base_audio_src_create):
	Improve latency query code.
	Use proper clock names.

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

	* tests/check/generic/states.c: (GST_START_TEST):
	  Copy the states.c test from core again
	* tests/check/Makefile.am:
	  ignore cdio and cdparanoiasrc

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2007-02-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index), (check_default),
	(audio_convert_prepare_context), (audio_convert_convert):
	  Also make valgrind happy and avoid copying data in some cases.

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2007-02-28  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/gstaudioconvert.c:
	(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
	(gst_audio_convert_transform_caps):
	* tests/check/elements/audioconvert.c: (GST_START_TEST),
	(audioconvert_suite):
	  Don't run inplace if that overwrites source data as we go. Add more
	  tests. Fixes #339837 even more.

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2007-02-27  Julien MOUTTE  <julien@moutte.net>

	* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
	(msg_segment_done): Fix various seeking bugs (Slider was not
	updating when doing a non flushing seek, Reverse playback 
	on segment seek was wrong).

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2007-02-26  Wim Taymans  <wim@fluendo.com>

	* tests/examples/seek/seek.c: (stop_seek):
	When we stop scrubbing, don't leave the pipeline PLAYING when we
	requested a PAUSED state.

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2007-02-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse date strings in vorbis comments that have an invalid (zero)
	  month or day (#410396).

	* tests/check/libs/tag.c: (GST_START_TEST):
	  Test case for the above.

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

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

	* configure.ac:
	* ext/alsa/Makefile.am:
	* gst/audiotestsrc/Makefile.am:
	  Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).

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2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c:
	  Improve docs: point out that the application needs to assist playbin
	  with buffering.

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2007-02-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	* tests/check/libs/utils.c: (missing_msg_check_getters):
	  Change GStreamer marker prefix in detail string from 'gstreamer.net'
	  to just 'gstreamer'. Document the caps string component of the
	  decoder/encoder detail a bit better, since not everyone will be
	  familiar with the GStreamer media type/caps system (but they better
	  enjoy nested itemized lists).

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2007-02-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/netbuffer/gstnetbuffer.c:
	(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
	  Fix copying of GstNetBuffer (would crash before, or at least lead to
	  invalid memory access, #410772), for now by copying the GstBuffer copy
	  code from the core over here so we can copy the GstBuffer fields on a
	  provided buffer instance (of type GstNetBuffer in this case). Would be
	  better to fix this with some support by the core though (and in the long
	  run change the broken GstBuffer/GstMiniObject copy semantics, #393099).

	* tests/check/Makefile.am:
	  Enable unit test for GstNetBuffer.

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2007-02-22  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_init): Disable pull-mode activation until we
	figure out how to make audio sinks go to PLAYING.

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2007-02-22  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	(double_hq), (audio_convert_get_func_index),
	(audio_convert_prepare_context), (audio_convert_convert):
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
	(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
	* gst/audioconvert/gstchannelmix.h:
	* tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Add float as an intermediate format, as well as float mixing. Enable
	  test that was failing before. Fixes #339837

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2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/examples/seek/seek.c: (do_seek):
	Undo the previous commit: -1 as a stop time implies that the stop
	time is the end of file, clearing any previously configured segment.

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2007-02-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/examples/seek/seek.c: (do_seek):
	Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.

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

	* gst/volume/gstvolume.c: (volume_process_int16),
	(volume_process_int16_clamp), (volume_set_caps):
	  Unbreak volume, value remains gint.

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

	* gst/volume/gstvolume.c: (volume_choose_func),
	(volume_update_real_volume), (gst_volume_set_volume),
	(gst_volume_init), (volume_process_double), (volume_process_float),
	(volume_process_int16), (volume_process_int16_clamp),
	(volume_set_caps), (volume_transform_ip), (volume_update_volume):
	* gst/volume/gstvolume.h:
	  Extend float audio support (double) and some int->uint cleanups.

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2007-02-20  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
	(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
	(sort_end_pads), (gst_decode_group_expose),
	(gst_decode_group_hide):
	Don't free groups from the streaming threads. Just put them aside and
	free them in dispose.

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2007-02-20  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (connect_element),
	(pad_added_group_cb), (gst_decode_group_check_if_blocked),
	(sort_end_pads), (gst_decode_group_expose):
	Handle dynamic pads within groups.
	Sort pads before exposing them in order to make playbin happy.
	There still is a race with the multiqueue filling up. This should be
	solved separately.
	Fixes #398721

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/base-utils.c:
	* gst-libs/gst/utils/descriptions.c:
	* gst-libs/gst/utils/install-plugins.c:
	* gst-libs/gst/utils/missing-plugins.c:
	  Some more docs (and descriptions for two subtitle formats).

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/audio.c:
	  Fix documentation.

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2007-02-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Yves Lefebvre  <ivanohe abacom com>

	* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
	  Don't leak caps. Fixes #408278.

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2007-02-15  Stefan Kost  <ensonic@users.sf.net>

	* ext/cdparanoia/gstcdparanoiasrc.h:
	* ext/ogg/gstoggdemux.h:
	* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
	(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
	(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/interfaces/videoorientation.h:
	* gst/adder/gstadder.h:
	  More docs coverage and some ChangeLog surgery (add missing names)

Wim Taymans's avatar
Wim Taymans committed
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2007-02-15  Wim Taymans  <wim@fluendo.com>

	* sys/ximage/ximagesink.c:
	(gst_ximagesink_calculate_pixel_aspect_ratio):
	* sys/xvimage/xvimagesink.c:
	(gst_xvimagesink_calculate_pixel_aspect_ratio):
	Small constifications.

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

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
	(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
	(gst_base_audio_sink_async_play),
	(gst_base_audio_sink_change_state):
	Answer latency query.
	Use configured latency when syncing.
	Fix clock slaving.

	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
	(gst_base_audio_src_query), (gst_base_audio_src_change_state):
	Fix possible memleak.
	Implement latency query.
	Small cleanups.

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

	* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
	Ignore errors in reset, these are not fatal. They also grab the element
	lock which is already taking when this function is called. Fixes
	#405451.

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

	* configure.ac:
	  Remove 'tests/examples/xerror/Makefile' from output files again.

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

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Also crossref against gst-plugins-base-libs.

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

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

	* gst-libs/gst/audio/audio.h:
	  Source formatting.

	* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
	  Add own debug category.

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

	Patch by: René Stadler <mail at renestadler de>

	* gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
	  (#403597).

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2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (setup_source):
	  When we have external subtitles and wait for the subtitle decodebin
	  to get up and running, we set up a (sync) bus handler for the
	  subtitle decodebin, so we can stop waiting when it posts an error
	  message. However, we should do that before we set the subtitle
	  decodebin's state to playing, otherwise things are racy and we might
	  miss error messages posted before we had a chance to set up the bus.
	  This should finally fix totem hanging on .txt pseudo-subtitle files.
	  
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2007-02-10  Sébastien Moutte  <sebastien at moutte dot net>

	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	  Use gst_gdouble_to_guint64 for conversions.
	* win32/common/config.h.in:
	  Add a define for GST_INSTALL_PLUGINS_HELPER
	* win32/common/libgstaudio.def:
	* win32/common/libgstcdda.def:
	* win32/common/libgstnetbuffer.def:
	* win32/common/libgstrtp.def:
	* win32/common/libgutils.def:
	  Add new exported functions.
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstnetbuffer.dsp:
	* win32/vs6/libgstplaybin.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstvorbis.dsp:
	* win32/vs6/libgstcdda.dsp:
	* win32/vs6/libgstgdp.dsp:
	* win32/vs6/libgstutils.dsp:
	  Update and add new project files.

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2007-02-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
	(subrip_remove_unhandled_tags), (parse_subrip):
	  For SubRip (.srt) subtitles, ignore all markup tags we don't
	  handle (like font tags, for example).

	* tests/check/elements/subparse.c:
	  Add test for this.

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

	* gst/playback/gstdecodebin.c: (add_fakesink),
	(gst_decode_bin_change_state):
	* gst/playback/gstdecodebin2.c: (add_fakesink),
	(gst_decode_bin_change_state):
Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
1272
	  Don't error out if there is no fakesink in the NULL to READY state
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	  change, since when decodebin is re-used, we're only adding the
	  fakesink element in READY to PAUSED.

	* tests/check/elements/decodebin.c:
	(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
	(decodebin_suite):
	  Minimal unit test to make sure we can use the same decodebin
	  instance twice (at least with audiotestsrc input).

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

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
	  Try to get devic-name from device string first, and from handle only
	  as fallback (seems to yield better results and is more robust
	  against buggy probing code on the application side).

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

	Based on patch by: Julien Puydt <julien.puydt at laposte net>

	* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
	(gst_alsa_find_device_name):
	* ext/alsa/gstalsa.h:
	* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
	* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
	  Improve device-name detection a bit, especially in the case where
	  the device is not actually open (#405020, #405024). Move common code
	  into gstalsa.c instead of duplicating it.

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

	* gst/audioconvert/gstaudioconvert.c:
	  Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.

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2007-02-06  Julien MOUTTE  <julien@moutte.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_clear),
	(gst_xvimagesink_interface_supported),
	(gst_xvimagesink_probe_get_properties),
	(gst_xvimagesink_probe_probe_property),
	(gst_xvimagesink_probe_needs_probe),
	(gst_xvimagesink_probe_get_values),
	(gst_xvimagesink_property_probe_interface_init),
	(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	(gst_xvimagesink_init), (gst_xvimagesink_class_init),
	(gst_xvimagesink_get_type):
	* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
	for XVAdaptors so that one can choose the adaptor to use with 
	gstreamer-properties.

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2007-02-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/audioconvert/gstaudioconvert.c:
	  Also mention that a conversion from double to float is suboptimal still.

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

	* gst-libs/gst/audio/gstaudiofilter.c:
	(gst_audio_filter_class_init), (gst_audio_filter_change_state):
	  Clear our formats structure and free the caps contained in it when
	  shutting down.

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2007-02-05  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_callback): Update basesink->offset so that we
	pull monotonically increasing offsets instead of, um, seeking back
	to 0 each time. Fixes alsasrc ! alsasink!

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2007-02-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videoscale/gstvideoscale.c:
	  A width and height of 1 makes us crash, so increase minimum size to
	  2x2 pixels until someone feels like fixing this (#404512).

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2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
	  Add small test to make sure request pads are cleaned up properly
	  even if oggmux never changes state out of NULL.

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2007-02-04  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/utils.c: (GST_START_TEST):
	  Fix unit test. Turns out things work much better when you
	  NULL-terminate string arrays. Should make p5 build bot happy again.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/gstaudiofiltertemplate.c:
	(gst_audio_filter_template_base_init),
	(gst_audio_filter_template_class_init),
	(gst_audio_filter_template_init),
	(gst_audio_filter_template_set_property),
	(gst_audio_filter_template_get_property),
	(gst_audio_filter_template_setup),
	(gst_audio_filter_template_filter),
	(gst_audio_filter_template_filter_inplace), (plugin_init):
	  Oops, forgot to commit fixed-up example.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
	(gst_audio_filter_class_init), (gst_audio_filter_init),
	(gst_audio_filter_set_caps),
	(gst_audio_filter_class_add_pad_templates):
	* gst-libs/gst/audio/gstaudiofilter.h:
	  Port GstAudioFilter to 0.10. This change technically breaks
	  API and ABI (and thus also every library developer's heart),
	  but seems justifiable on the grounds that the base class was
	  completely unusable before (ie. would crash immediately when
	  actually used). Fixes #403963 (and eventually also #403572).
	  Also document all of this a bit.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/install-plugins.c:
	(gst_install_plugins_spawn_child):
	* tests/check/libs/utils.c:
	(test_base_utils_install_plugins_do_callout):
	  Lowering log level to see why things fail on the p5 build bot;
	  fix some typos in unit test messages.

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2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/libs/utils.c:
	(test_base_utils_install_plugins_do_callout):
	  Don't hard-code temp directory for test helper; use GLib functions
	  to write out file and do error checking etc.

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

	* gst-libs/gst/utils/Makefile.am:
	* gst-libs/gst/utils/base-utils.h:
	* gst-libs/gst/utils/install-plugins.c:
	(gst_install_plugins_context_set_xid),
	(gst_install_plugins_context_new),
	(gst_install_plugins_context_free),
	(gst_install_plugins_get_helper),
	(gst_install_plugins_spawn_child),
	(gst_install_plugins_return_from_status),
	(gst_install_plugins_installer_exited),
	(gst_install_plugins_async), (gst_install_plugins_sync),
	(gst_install_plugins_return_get_name),
	(gst_install_plugins_installation_in_progress):
	* gst-libs/gst/utils/install-plugins.h:
	  API: add API for applications to initiate installation of missing
	  plugins, ie. gst_install_plugins_async() primarily.
	  Based on libgimme-codec by Ryan Lortie.

	* configure.ac:
	  Add --with-install-plugins-helper configure option so distros can specify
	  the path of the helper script or program to call when plugin installation
	  is requested (distros: please do any argument munging in this helper
	  script instead of patching GStreamer to pass arguments differently
	  to another program directly).

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Build and document new API.

	* tests/check/libs/utils.c: (result_cb),
	(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
	(libgstbaseutils_suite):
	  Some simple checks for the new API.

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

	* tests/check/elements/audioconvert.c: (test_float_conversion):
	  Add small test for 32bit float <=> 64bit float conversion (works
	  only one way so far, 32=>64 produces structured noise).

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

	* gst/audioconvert/gstaudioconvert.c:
	(set_structure_widths_32_and_64), (make_lossless_changes):
	  We don't support floats with a width of 40, 48 or 56 bits.

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

	* gst/audioconvert/audioconvert.c: (float), (double),
	(audio_convert_get_func_index):
	* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	(make_lossless_changes):
	  Support for 64-bit float audio in audioconvert (#339837)

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2007-02-01  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Holger Wansing  <linux wansing-online de>

	* po/LINGUAS:
	* po/de.po:
	  Add German translation (#352069).

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2007-02-01  Sebastian Dröge  <slomo@circular-chaos.org>

	reviewed by: Wim Taymans <wim@fluendo.com>

	* ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
	(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
	Use newly added GstCollectPads API to free the allocated resources in
	the GstOggPad structures (#402393).

1479 1480 1481 1482 1483 1484 1485 1486 1487
2007-01-31  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/playback/gstplaybin.c: (gen_vis_element):
	  Add audioresample+audioconvert in front of the visualisation
	  element, so that elements like libvisual 0.4 that don't support all
	  samplerates can work.

	  Fixes: #402505

1488 1489 1490 1491 1492 1493 1494 1495
2007-01-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
	(gst_play_base_bin_get_streaminfo_value_array):
	  Take some locks and make a copy of the streaminfo value array we
	  maintain while holding the lock, so that the application can
	  retrieve the stream-info as a value array in a thread-safe way.

1496 1497 1498 1499 1500
2007-01-30  Wim Taymans  <wim@fluendo.com>

	* gst/audioconvert/gstaudioconvert.c:
	Don't fail on 0 sized buffers. Fixes #396835.

1501 1502 1503 1504 1505 1506
2007-01-29  David Schleef  <ds@schleef.org>

	* gst/typefind/gsttypefindfunctions.c:
	  Detect BBCD as video/x-dirac, so we can play raw dirac
	  streams.

1507 1508 1509 1510 1511 1512 1513 1514
2007-01-29  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/theora/theoraenc.c: (theora_enc_chain):
	  Check return value of theora_encode_header(), or we might try to
	  allocate a random number of bytes. theora_encode_header() can fail
	  if libtheora has been compiled with encoding support disabled.
	  Fixes #398110.

1515 1516 1517 1518 1519
2007-01-29  Wim Taymans  <wim@fluendo.com>

	* tests/check/gst/.cvsignore:
	Do as buildbot says.

1520 1521 1522 1523 1524 1525 1526 1527
2007-01-29  Wim Taymans  <wim@fluendo.com>

	* ext/libvisual/visual.c: (gst_visual_src_setcaps):
	Fix strides in libvisual. Gst uses X strides.
	Inspired by: <ed at catmur dot co dot uk> and 
	<tim at centricular dot net>
	Fixes #401118.

1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543
2007-01-27  Wim Taymans  <wim@fluendo.com>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	(gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
	(gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
	(gst_ogg_demux_perform_seek),
	(gst_ogg_demux_bisect_forward_serialno),
	(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
	(gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
	(gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
	(gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
	* ext/ogg/gstoggdemux.h:
	Properly propagate streaming errors when we are scanning the file for
	chains so that we don't crash when shut down. Might fix some crashers
	when quickly switching oggs in RB such as #332503 and #378436.

1544 1545 1546 1547 1548 1549
2007-01-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	  Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
	  error code as well.

1550 1551 1552 1553 1554
2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstplaybasebin.c: (remove_source):
	Don't try to disconnect a signal from a finalized object.

1555 1556 1557 1558 1559 1560 1561
2007-01-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
	  Cast lock macro parameters to make sure we're actually accessing the
	  lock member at the right class level. Free list itself in _dispose()
	  as well and NULL it in case dispose gets called multiple times.

1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572
2007-01-25  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c:
	(gst_decode_bin_dispose),(gst_decode_bin_finalize):
	Free GstDecodeGroups no longer used.
	(gst_decode_group_expose):
	Don't unlock too many times !
	(deactivate_free_recursive):
	Free iterator once we're done with it.
	Fix for recursively deactivating elements (stop at ghostpads).

1573 1574 1575 1576 1577 1578 1579
2007-01-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (handoff):
	  Fix up caps on the frame buffer before we save it and potentially
	  make it accessible to other threads via g_object_get; also use
	  gst_buffer_replace() instead of gst_mini_object_replace().

1580 1581 1582 1583 1584
2007-01-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Make getting the current frame thread-safe.

1585 1586 1587 1588 1589 1590 1591
2007-01-25  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
	(gst_decode_group_new), (gst_decode_group_free):
	Set queues to bigger sizes to cope with HD contents.
	Fix some mutex freeing and add comment about MT safe methods.

1592 1593 1594 1595 1596 1597 1598 1599 1600 1601
2007-01-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
	(gst_text_overlay_text_event):
	  Don't unnecessarily ref (and then leak) upstream events if the text
	  pad is not linked. Fixes #399948.

	* tests/check/gst-plugins-base.supp:
	  Add suppression for pango on edgy/x86 for textoverlay test.

1602 1603 1604 1605 1606
2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst-libs/gst/rtp/gstrtpbuffer.h:
	Add some more fixed payloads.

1607 1608 1609 1610 1611 1612
2007-01-23  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	  Error out properly if we get an error from libogg while reading the
	  BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).

1613 1614 1615 1616 1617 1618 1619 1620 1621 1622 1623 1624 1625 1626
2007-01-23  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
	  Don't leak mutex.

	* tests/check/elements/playbin.c:
	(test_sink_usage_video_only_stream),
	(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
	(test_suburi_error_wrongproto), (test_missing_urisource_handler),
	(test_missing_suburisource_handler),
	(test_missing_primary_decoder), (playbin_suite):
	  Run all tests once with decodebin and once with decodebin2.
	  One test does not pass yet with decodebin2.

1627 1628 1629 1630 1631 1632 1633 1634
2007-01-23  Edward Hervey  <edward@fluendo.com>

	* ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
	Fix the cases where oggmux doesn't properly figure out that all
	sinkpads have gone EOS, and therefore doesn't push out the remaining
	buffers and the final EOS event.
	Fixes #363379

1635 1636 1637 1638 1639 1640 1641
2007-01-23  Julien MOUTTE  <julien@moutte.net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	Don't lock on navigation event push, just on keysym to string.
	Fixes #397673 again.

1642 1643 1644 1645 1646 1647 1648 1649 1650 1651 1652
2007-01-22  Edward Hervey  <edward@fluendo.com>

	* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
	(get_current_group), (group_demuxer_event_probe),
	(gst_decode_group_expose), (deactivate_free_recursive),
	(gst_decode_group_free):
	Cleanups.
	Don't forget to emit 'no-more-pads' once a group is exposed.
	Cleanup elements from a DecodeGroup once we remove it.
	Protect call to gst_decode_group_expose() with the decodebin lock.

1653 1654 1655 1656 1657 1658 1659 1660
2007-01-22  Julien MOUTTE  <julien@moutte.net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	Looking at Xorg code i can't figure out if that XKeysymToString
	function is thread sensible or not. Lock it just in case as
	recommended by Radek Doulik <rodo at ximian dot com>.

1661 1662 1663 1664 1665 1666
2007-01-22  Julien MOUTTE  <julien@moutte.net>

	* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	Lock that X Call as well. Fixes #397673.

1667 1668 1669 1670 1671 1672 1673 1674 1675 1676 1677 1678
2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
	  Don't go into an endless loop if the file starts with 00 00 01 2X,
	  like quicktime redirect files might. Fixes #396042.

	* tests/check/Makefile.am:
	* tests/check/gst/.cvsignore:
	* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
	(typefindfunctions_suite):
	  Add unit test for the above.

1679 1680 1681 1682 1683
2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  On second thought, use "depth" field rather than "bpp" field.

1684 1685 1686 1687 1688
2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Camtasia caps apparently need a bpp field (#398875).

1689 1690 1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701 1702
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (setup_subtitle),
	(gen_source_element), (gst_play_base_bin_change_state):
	  Attempt at a better error message in case we don't have the required
	  URI handler installed; post missing-plugin message also when we're
	  missing an URI handler for the subtitle URI; clean up properly also
	  when an error occurs and we never made it to PAUSED state.

	* tests/check/elements/playbin.c: (GST_START_TEST),
	(playbin_suite):
	  Check that we're also getting a missing-plugin messsage for a
	  missing subtitle URI handler (and clean up properly).

1703 1704 1705 1706 1707
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
	  Plug a few reference leaks.

1708 1709 1710 1711 1712 1713
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  Lower probability a bit if the marker isn't right at the start,
	  to decrease the chance of false positives.

1714 1715 1716 1717 1718 1719 1720
2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  Small mpeg2 system stream typefinding improvement: make typefinder
	  probe a bit into the stream instead of just looking for a marker
	  at the beginning. Fixes #397810.

1721 1722 1723 1724 1725
2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/audioconvert/gstchannelmix.c:
	  Remove compatibility cruft for prehistoric GLib versions.

1726 1727 1728 1729 1730 1731 1732 1733 1734 1735 1736 1737 1738
2007-01-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/Makefile.am:
	* gst/playback/gstdecodebin.c: (close_pad_link):
	* gst/playback/gstdecodebin2.c: (analyze_new_pad):
	* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	(gst_play_base_bin_handle_message_func), (unknown_type):
	  Let decodebin be the element to post missing-plugin messages for
	  missing decoders (rather than playbin); make playbin implement
	  GstBin::handle_message so we can suppress missing-plugin messages
	  for types we're not handling on purpose (don't want to bring up an
	  installer in those cases).

1739 1740 1741 1742 1743 1744 1745 1746
2007-01-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	* gst-libs/gst/tag/gstvorbistag.c:
	(gst_tag_list_to_vorbiscomment_buffer):
	* gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
	  Fix potentially unaligned access (#397207).

1747 1748 1749 1750 1751 1752 1753 1754 1755
2007-01-16  Stefan Kost  <ensonic@users.sf.net>

	* tests/examples/seek/seek.c: (set_scale), (update_scale),
	(do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
	(rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
	(main):
	  Allow to toggle looping while it plays. Fix callback prototype. Clean
	  up code a bit more. Add copyright header.

1756 1757 1758 1759 1760
2007-01-16  Stefan Kost  <ensonic@users.sf.net>

	* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
	  Red and blue mask was swapped (spotted by Dan Williams).

1761 1762 1763 1764 1765 1766
2007-01-15  Stefan Kost  <ensonic@users.sf.net>

	* gst-libs/gst/tag/gstid3tag.c:
	* gst-libs/gst/tag/gstvorbistag.c:
	  Use new beats-per-minute tag from core.

1767 1768 1769 1770 1771 1772
2007-01-15  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Add new files with translatable strings, so they actually make it
	  into the template file one day.

1773 1774 1775 1776 1777 1778 1779 1780 1781 1782 1783 1784 1785 1786 1787
2007-01-12  Andy Wingo  <wingo@pobox.com>

	* gst-libs/gst/audio/gstbaseaudiosink.c
	(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
	(gst_base_audio_sink_activate_pull): Remove the handwavey nego
	stuff, as the base class handles this now. Actually tell the ring
	buffer to start.
	(gst_base_audio_sink_callback): Cast the ring buffer correctly.
	How did this work before? Maybe I'm not as awesome a programmer as
	I think.

	* gst-libs/gst/audio/gstbaseaudiosrc.c
	(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
	of a pad function.

1788 1789 1790 1791 1792 1793
2007-01-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
	  Remove more fields so that the application can better blacklist
	  formats that have been tried before.

1794 1795 1796 1797 1798 1799
2007-01-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst-libs/gst/audio/mixerutils.h:
	  Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
	  used when compiling with c++ compilers as well.

1800 1801 1802 1803 1804
2007-01-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/typefind/gsttypefindfunctions.c:
	  Fix comment.

1805 1806 1807 1808 1809 1810 1811 1812
2007-01-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/playback/gstplaybin.c: (post_missing_element_message),
	(gen_video_element), (gen_text_element), (gen_audio_element),
	(gen_vis_element):
	  Post missing-plugin messages also when we error out because
	  converters, textoverlay or auto*sinks are missing (#161922).

1813 1814 1815 1816 1817 1818 1819 1820 1821 1822
2007-01-10  Wim Taymans  <wim@fluendo.com>

	* gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
	(is_demuxer_element), (new_caps):
	* gst/playback/gstplaybasebin.c: (source_new_pad):
	Fix the case where we try to ref a NULL element when we delay a link
	because of unfixed caps.
	Set the state of autoplugged decodebins to PAUSED.
	RT