Release 0.10.16

Original commit message from CVS:
Release 0.10.16

ChangeLog

@@ -1,3 +1,10 @@
+=== release 0.10.16 ===
+
+2008-01-28  Jan Schmidt <jan.schmidt@sun.com>
+
+	* configure.ac:
+	  releasing 0.10.16, "Scheduled Interruption"
+
 2008-01-22  Wim Taymans  <wim.taymans@collabora.co.uk>
 
 	Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>

NEWS

@@ -1,9 +1,71 @@
-This is GStreamer Base Plug-ins 0.10.15, "No need to argue"
+This is GStreamer Base Plug-ins 0.10.16, "Scheduled Interruption"
 
-Please note that decodebin2 API included in this release is still
-considered unstable and WILL change in future releases. At this stage, only
-developers or early adopters should consider using the decodebin2 API embodied
-in its signals and properties.
+IMPORTANT NOTES
+
+1) Please note that decodebin2 and playbin2 API included in this release is
+still considered unstable and WILL change in future releases. At this stage,
+only developers or early adopters should consider using decodebin2 or playbin2
+API embodied in their signals and properties.
+
+2) On some systems, the current release of gst-plugins-good (0.10.6) may fail to
+build against this release of gst-plugins-base with an error like:
+
+gstid3v2mux.cc:547: error: 'GST_TAG_MUSICBRAINZ_SORTNAME' was not declared in this scope
+
+In this case, you should either patch the configure file of gst-plugins-good to
+remove -DGST_DISABLE_DEPRECATED from DEPRECATED_CFLAGS=, or else compile
+with make DEPRECATED_CFLAGS=''
+
+3) Some users may experience problems using the 'mp3parse' element from the
+previous gst-plugins-ugly release (0.10.6). This is due to a bug in mp3parse
+exposed by changes in decodebin in gst-plugins-base. It will be fixed in the
+upcoming release of gst-plugins-ugly next month. In the meantime as a
+workaround, you can set the rank of mp3parse to GST_RANK_NONE in 
+gst-plugins-ugly/gst/mpegaudioparse/gstmpegaudioparse.c when compiling, or
+or remove the /usr/lib/gstreamer-0.10/libgstmpegaudioparse.so file entirely.
+
+Changes since 0.10.15:
+
+      * Handle newer Theora granule-pos semantics
+      * Introducing first alpha version playbin2 - the upcoming successor to
+	playbin
+      * Fixes in playbin handling of stream-switching
+      * New API for uniform handling of raw-video format buffers.
+      * Improvements for RTSP/RTP handling
+      * RIFF lib additions for VC-1 and AVC1 fourccs
+      * Many other bug-fixes and improvements
+
+Bugs fixed since 0.10.15:
+
+      * 506132 : Review of changes in video/video.h
+      * 320984 : [oggdemux] cannot handle multiple chains
+      * 373011 : [playbin] throws error when switching off subtitles
+      * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
+      * 462740 : [streamselector] patch to improve default stream selection
+      * 486840 : [alsamixer] use _all variants when setting the mixer
+      * 497964 : theoraenc test fails
+      * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
+      * 499697 : Provide better pkg-config files 
+      * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
+      * 503440 : The control sockets used by gstrtspconnection.c are never...
+      * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
+      * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
+      * 508138 : [decodebin] does not error out if pad activation fails
+      * 509762 : missing file in win32/MANIFEST
+      * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
+      * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
+      * 496761 : [PATCH] RTSP message leaks memory when uninitialized
+      * 500763 : SIGSEGV while playing ogg audio file
+
+API additions since 0.10.15:
+
+      * New GstVideoFormat API and helper functions in libgstvideo
+      * gst_base_audio_sink_set_provide_clock()
+      * gst_base_audio_sink_get_provide_clock()
+      * gst_base_audio_sink_set_slave_method()
+      * gst_base_audio_sink_get_slave_method()
+      * gst_base_audio_src_set_provide_clock()
+      * gst_base_audio_src_get_provide_clock()
 
 Changes since 0.10.14:
 

RELEASE

@@ -1,5 +1,5 @@
 
-Release notes for GStreamer Base Plug-ins 0.10.15 "No need to argue"
+Release notes for GStreamer Base Plug-ins 0.10.16 "Scheduled Interruption"
 
 
 
@@ -54,89 +54,47 @@ contains a set of less supported plug-ins that haven't passed the
 
 Features of this release
 
-      * RTP/RTSP/RTCP/SDP support improved
-      * New FFT support library libgstfft, based on Kiss FFT
-      * New formats supported in volume and audiotestsrc
-      * Fixes in audiorate and videorate
-      * Audio capture fixes
-      * Playbin and decodebin fixes
-      * New tagdemux base class for ID3/APE style tag readers
-      * Fix a nasty crash in the X sinks on shutdown
-      * New tags supported
-      * Add support for multichannel WAV files.
-      * Preserve channel layout information when up/down-mixing.
-      * Many bug-fixes and improvements
-      * 
+      * Handle newer Theora granule-pos semantics
+      * Introducing first alpha version playbin2 - the upcoming successor to playbin
+      * Fixes in playbin handling of stream-switching
+      * New API for uniform handling of raw-video format buffers.
+      * Improvements for RTSP/RTP handling
+      * RIFF lib additions for VC-1 and AVC1 fourccs
+      * Many other bug-fixes and improvements
 
 Bugs fixed in this release
 
-      * 475395 : decodebin2 leaks request-pads
-      * 475451 : [decodebin2] leaks ghostpad
-      * 378770 : [xvimagesink] race condition in event thread?
-      * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
-      * 430677 : [audioconvert] does not preserve channel positions when f...
-      * 442654 : [volume] controller bypassed by default
-      * 445529 : [volume] support for 24/32-bit audio/x-raw-int
-      * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
-      * 451970 : Subparse requires HTML parser
-      * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
-      * 459334 : [textoverlay] expose pango line alignment property
-      * 459585 : [basertpdepayload] api without namespace
-      * 460422 : [audiotestsrc] Add support for float and double output
-      * 462805 : [alsa] compilation fails with gcc 4.2
-      * 462979 : Add 'silent' property to GstTimeOverlay
-      * 463215 : [audioconvert] compile errors
-      * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
-      * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
-      * 464690 : Add connection-speed property to uridecodebin element
-      * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
-      * 465028 : some warnings with mingw
-      * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
-      * 468129 : [basertpaudiopayload] event handler returns the wrong value
-      * 468619 : New library gstfft: FFT library for integer and float typ...
-      * 470456 : [API] add gst_missing_*_installer_detail_new()
-      * 470766 : [ssaparse] line breaks in SSA subtitle parser
-      * 471067 : Make the SDP code useable for generating SDP descriptions
-      * 471194 : [rtpbuffer] RTP headers are wrong for win32
-      * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
-      * 474384 : gstrtsp-enumtypes.c and .h needed for win32
-      * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
-      * 475731 : rtspconnection is able to read incomplete messages
-      * 483620 : All Rtp buffers are discarded --  gst_rtp_buffer_get_payl...
-      * 484989 : memleak, not unrefed caps for gstbasertppayload.c
-      * 489010 : Please change default channel order for WAVE_EXT-less .wa...
-      * 491722 : [playbin] regression: crash with external subtitles
-      * 492098 : [GstFFT] Broken scaling
-      * 492114 : Build issues on Windows/MSVC
-      * 492306 : compilation errors with MinGW
-      * 492813 : Missing symbols in libgstrtp.def
-      * 493986 : Build issues on Windows (missing symbols)
-      * 494346 : pre-release vs6 patch
-      * 496548 : Including malloc.h breaks macos build
-      * 496724 : DSW file references non-existent DSP files
-      * 464079 : audiotestsrc doesn't respond to conversion queries properly
-      * 442065 : floatcast.h includes config.h and might break other apps
-      * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
-      * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
-      * 464028 : Move connection-speed from playbin to playbasebin
+      * 506132 : Review of changes in video/video.h
+      * 320984 : [oggdemux] cannot handle multiple chains
+      * 373011 : [playbin] throws error when switching off subtitles
+      * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
+      * 462740 : [streamselector] patch to improve default stream selection
+      * 486840 : [alsamixer] use _all variants when setting the mixer
+      * 497964 : theoraenc test fails
+      * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
+      * 499697 : Provide better pkg-config files
+      * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
+      * 503440 : The control sockets used by gstrtspconnection.c are never...
+      * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
+      * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
+      * 508138 : [decodebin] does not error out if pad activation fails
+      * 509762 : missing file in win32/MANIFEST
+      * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
+      * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
+      * 496761 : [PATCH] RTSP message leaks memory when uninitialized
+      * 500763 : SIGSEGV while playing ogg audio file
 
 API changed in this release
 
 - API additions:
 
-* GstTagDemux base class for simple tag demuxers
-* GstBaseAudioSrc::provide-clock property
-* gst_rtcp_ntp_to_unix()
-* gst_rtcp_unix_to_ntp()
-* gst_rtp_buffer_get_header_len()
-* gst_rtp_buffer_get_extension_data()
-* gst_rtp_buffer_compare_seqnum()
-* gst_rtp_buffer_ext_timestamp()
-* gst_rtcp_packet_sdes_copy_entry()
-* gst_install_plugins_supported()
-* gst_missing_*_installer_detail_new() convenience API
-* gst_rtsp_connection_poll()
-* GstTextOverlay::line-alignment property
+* New GstVideoFormat API and helper functions in libgstvideo
+* gst_base_audio_sink_set_provide_clock()
+* gst_base_audio_sink_get_provide_clock()
+* gst_base_audio_sink_set_slave_method()
+* gst_base_audio_sink_get_slave_method()
+* gst_base_audio_src_set_provide_clock()
+* gst_base_audio_src_get_provide_clock()
 
 Download
 
@@ -166,40 +124,22 @@ Applications
 
 Contributors to this release
 
-      * Stefan Kost
-      * Alexander Shopov
-      * Damien Lespiau
-      * Dan Williams
-      * Daniel Díaz
+      * Bastien Nocera
       * David Schleef
-      * Davyd Madeley
-      * Funda Wang
-      * Haakon Sporsheim
-      * Ilkka Tuohela
-      * Jakub Bogusz
+      * Edward Hervey
       * Jan Schmidt
-      * Jason Kivlighn
-      * Jens Granseuer
-      * Johan Dahlin
-      * Jorge González González
-      * Josep Torra Valles
+      * Jerone Young
+      * Joe Peterson
       * Julien MOUTTE
-      * Laurent Glayal
+      * Julien Moutte
       * Michael Smith
-      * Mogens Jaeger
-      * Ole André Vadla Ravnås
-      * Olivier Crete
       * Peter Kjellerstedt
-      * Renato Filho
-      * René Stadler
+      * Robin Stocker
       * Sebastian Dröge
       * Sebastien Moutte
       * Stefan Kost
       * Thijs Vermeir
-      * Thomas Vander Stichele
       * Tim-Philipp Müller
       * Tommi Myöhänen
-      * Vincent Torri
       * Wim Taymans
-      * Yang Hong
 

configure.ac

@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
 dnl initialize autoconf
 dnl releases only do -Wall, cvs and prerelease does -Werror too
 dnl use a three digit version number for releases, and four for cvs/prerelease
-AC_INIT(GStreamer Base Plug-ins, 0.10.15.1,
+AC_INIT(GStreamer Base Plug-ins, 0.10.16,
     http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
     gst-plugins-base)
 
@@ -44,7 +44,7 @@ dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
 dnl - interfaces added -> increment AGE
 dnl - interfaces removed -> AGE = 0
 dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 11, 0, 11)
+AS_LIBTOOL(GST, 12, 0, 12)
 
 dnl FIXME: this macro doesn't actually work;
 dnl the generated libtool script has no support for the listed tags.