ChangeLog 1.22 MB
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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
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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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	* 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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	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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	* 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

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

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

	* configure.ac:
	Back to CVS

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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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	* 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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	* 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.

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

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
	(gst_rtsp_message_init_response),
	(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
	(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
	(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	(gst_rtsp_message_get_body), (dump_key_value):
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse):
	* gst-libs/gst/rtsp/gstrtsprange.h:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst-libs/gst/rtsp/gstrtspurl.c:
	Fix some more RTSP docs.
	Add some missing methods for dealing with messages.

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2007-07-24  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/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	(gst_rtsp_connection_connect), (add_auth_header),
	(gst_rtsp_connection_write), (gst_rtsp_connection_send),
	(read_body), (gst_rtsp_connection_receive),
	(gst_rtsp_connection_next_timeout),
	(gst_rtsp_connection_reset_timeout),
	(gst_rtsp_connection_set_auth):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse):
	* gst-libs/gst/rtsp/gstrtspurl.h:
	Added beginnings of RTSP documentation.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/sdp/gstsdp.h:
	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
	(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
	(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val),
	(gst_sdp_message_add_attribute), (gst_sdp_media_new),
	(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
	(gst_sdp_media_get_media), (gst_sdp_media_set_media),
	(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
	(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
	(gst_sdp_media_get_format), (gst_sdp_media_add_format),
	(gst_sdp_media_get_information), (gst_sdp_media_set_information),
	(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
	(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
	(gst_sdp_media_set_key), (gst_sdp_media_get_key),
	(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
	(gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
	(print_media), (gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	Document the SDP library.
	Add some of the missing SDPMedia methods.

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2007-07-24  Wim Taymans  <wim.taymans@gmail.com>

	* configure.ac:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	* gst-libs/gst/rtsp/gstrtspbase64.h:
	* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
	(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
	(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
	(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
	(parse_response_status), (parse_request_line), (parse_line),
	(gst_rtsp_connection_read), (read_body),
	(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
	(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
	(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
	(gst_rtsp_connection_set_auth):
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
	(gst_rtsp_strresult), (gst_rtsp_method_as_text),
	(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
	(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
	(gst_rtsp_find_method):
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	(gst_rtsp_message_new), (gst_rtsp_message_init),
	(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
	(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
	(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
	(gst_rtsp_message_free), (gst_rtsp_message_add_header),
	(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
	(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
	(gst_rtsp_message_dump):
	* gst-libs/gst/rtsp/gstrtspmessage.h:
	* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	(parse_npt_range), (parse_clock_range), (parse_smpte_range),
	(gst_rtsp_range_parse), (gst_rtsp_range_free):
	* gst-libs/gst/rtsp/gstrtsprange.h:
	* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
	(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
	(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
	(range_as_text), (rtsp_transport_mode_as_text),
	(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
	(gst_rtsp_transport_free):
	* gst-libs/gst/rtsp/gstrtsptransport.h:
	* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
	(gst_rtsp_url_free), (gst_rtsp_url_set_port),
	(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
	* gst-libs/gst/rtsp/gstrtspurl.h:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/sdp/gstsdp.h:
	* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
	(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
	(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
	(gst_sdp_attribute_init), (gst_sdp_message_new),
	(gst_sdp_message_init), (gst_sdp_message_uninit),
	(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
	(gst_sdp_media_uninit), (gst_sdp_media_free),
	(gst_sdp_message_set_origin), (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_set_key),
	(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	(gst_sdp_message_get_attribute_val),
	(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
	(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
	(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
	(gst_sdp_media_get_attribute_val_n),
	(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
	(read_string), (read_string_del), (gst_sdp_parse_line),
	(gst_sdp_message_parse_buffer), (print_media),
	(gst_sdp_message_dump):
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	Move SDP and RTSP from helper objects in -good to a reusable library.
	Use a proper gst_ namespace.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
	(vorbis_dec_flush_decode):
	Use the new buffer clipping function from gstaudio here.

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2007-07-23  Sebastian Dröge  <slomo@circular-chaos.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
	* gst-libs/gst/audio/audio.h:
	* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
	API: Add buffer clipping function for raw audio buffers. Fixes #456656.
	Also add deprecation guards for gst_audio_structure_set_int() to the
	header.

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	* docs/libs/gst-plugins-base-libs-sections.txt:
	  Cleanup the docs.

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

	Patch by: Dan Williams <dcbw at redhat dot com>

	* gst/playback/gstplaybasebin.c:
	(gst_play_base_bin_get_streaminfo_value_array):
	Don't return NULL when querying the stream info value array but instead
	return an empty array. Fixes #459204.

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	* gst/playback/gsturidecodebin.c:
	  Init debug category before using it.

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2007-07-21  Jan Schmidt  <thaytan@noraisin.net>

	* gst-libs/gst/interfaces/mixer.h:
	Add padding vars in place of the signal pointers
	when building with DISABLE_DEPRECATED so that the
	interface structure doesn't change size.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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	* gst/audioconvert/audioconvert.c:
	Include math.h to fix compilation.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* po/POTFILES.in:
	  Add more files with translatable strings (#450875).

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

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

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

	* MAINTAINERS:
	Updating all the maintainers files

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

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

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

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

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

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

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

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

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

	* po/vi.po:
	  Update translations.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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	* configure.ac:
	check for large file support.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
	(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
	(gst_multi_fd_sink_remove_flush),
	(gst_multi_fd_sink_remove_client_link),
	(gst_multi_fd_sink_handle_client_write),
	(gst_multi_fd_sink_handle_clients):
	* gst/tcp/gstmultifdsink.h:
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2007-06-05  Wim Taymans  <wim@fluendo.com>

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

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

	* configure.ac:
	  Back to CVS

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=== release 0.10.13 ===

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

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

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

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

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

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	* gst-libs/gst/floatcast/floatcast.h:
	Define inline when needed on win32 builds. Fixes: #441295
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	Patch by: Sebastien Moutte  <sebastien@moutte.net>
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2007-05-29  Wim Taymans  <wim@fluendo.com>

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

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

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

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

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

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

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

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

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

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

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

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

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

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	Patch by: Mark Nauwelaerts <manauw at skynet be>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	Fixes: #393975

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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2069
2070
2007-05-12  Wim Taymans  <wim@fluendo.com>

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

2071
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2077
2007-05-12  Wim Taymans  <wim@fluendo.com>

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

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2007-05-11  Jan Schmidt  <thaytan@mad.scientist.com>

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

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

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

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

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2105
2007-05-10  Wim Taymans  <wim@fluendo.com>

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

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

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

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2007-05-09  Michael Smith <msmith@fluendo.com>

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

2121
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2007-05-09  Wim Taymans  <wim@fluendo.com>

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

2127
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2007-05-08  Stefan Kost  <ensonic@users.sf.net>

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

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2007-05-07  Michael Smith <msmith@fluendo.com>

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

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2007-05-04  Stefan Kost  <ensonic@users.sf.net>

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

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2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-05-04  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-05-04  Stefan Kost  <ensonic@users.sf.net>

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

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

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

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2007-05-03  Michael Smith <msmith@fluendo.com>

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

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2007-05-03  Edward Hervey  <edward@fluendo.com>

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

2204
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2209
2007-05-03  Edward Hervey  <edward@fluendo.com>

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

2210
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2007-05-01  Tim-Philipp Müller  <tim at centricular dot net>

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

2216
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2007-04-29  Wim Taymans  <wim@fluendo.com>

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

2221
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2007-04-27  Julien MOUTTE  <julien@moutte.net>

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

2230
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2007-04-27  Wim Taymans  <wim@fluendo.com>

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

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2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

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

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2007-04-25  Wim Taymans  <wim@fluendo.com>

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

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

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

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

2274
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2007-04-24  Sebastian Dröge  <slomo@circular-chaos.org>

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

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2007-04-24  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Dan Williams <dcbw redhat com>

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

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2007-04-23  Stefan Kost  <ensonic@users.sf.net>

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

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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).

2310
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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.

2319
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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).

2331
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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.

2336
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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

2357
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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).

2365
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2007-04-18  Stefan Kost  <ensonic@users.sf.net>

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

2370
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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.

2375
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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.

2410
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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.

2417
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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.

2424
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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.

2443
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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.