Release 1.1.1

sailfishos-mirror · Jun 5, 2013 · ff5d331 · ff5d331
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diff --git a/ChangeLog b/ChangeLog
diff --git a/NEWS b/NEWS
@@ -1,51 +1,2 @@
-This is GStreamer Base Plugins 1.0.2
+This is GStreamer Base Plugins 1.1.1
 
-Changes since 1.0.1:
-
-      * Parallel installability with 0.10.x series
-      * alsa: fix probing of supported formats, and advertise non-native-endianness formats as fallback
-      * audiobasesink: properly handle GAP events (fixing some isses with e.g. certain DVD menus)
-      * audioconvert: try harder to not convert or to preserve input format precision
-      * audiodecoder: leak fixes and refcounting fixes
-      * audioresample: re-enable the SSE/SSE2 code paths for better performance
-      * riff: fix paletted RGB formats and msvideo mapping
-      * rtsp: make formatting and parsing of range floating-point values locale-independent
-      * playbin: streamsynchronizer fixes, esp. for handling corner-cases near EOS
-      * tcpserver{sink,src}: add 'current-port' property and signal actually used port
-      * videoconvert: fix handling of paletted RGB formats
-      * videodecoder: don't leak message strings when error is not fatal
-      * videodecoder: finetune missing timestamp estimating
-      * videotestsrc: add palette for paletted RGB formats
-      * vorbistag: fix writing of image tags into vorbis comments
-
-Bugs fixed since 1.0.1:
-
-      * 580093 : tcpserversink,src: add 'current-port' property and signal actually used port when port=0 was set
-      * 680904 : alsasink: no supported formats detected with using USB sound card on big-endian system
-      * 683098 : videodecoder: log failure message if acquire_buffer failed
-      * 684411 : rtsp: range in SDP formatted according to locale
-      * 685273 : Pre-rolling on GAP events doesn't work properly for audio sinks
-      * 685711 : audio, video: update docs for virtual functions that require chaining up
-      * 685938 : [decodebin] Issues with group switching algorithm
-      * 686081 : adder: all unit tests crash now after collectpads changes
-      * 686298 : Cannot decode some AVI files with Microsoft Video 1
-
-Changes since 1.0.0:
-
-      * videodecoder and -encoder timestamp handling improvements
-      * thread-safey fixes for GstMeta registrations and GstVideoDecoder
-      * bug fixes
-
-Bugs fixed since 1.0.0:
-
-      * 684424 : playbin: external subtitles break playback
-      * 684832 : videodecoder: Takes stream lock in query function
-      * 685110 : encodebin fails to release mux request sink pad for GstId3Mux as it is a static one for this mux
-      * 685242 : rtsp: mark url argument of gst_rtsp_url_parse as out
-      * 685332 : GstMeta registry race
-      * 685490 : audioencoder: don't require base class to implement to start vfunc
-
-New features in 1.0.0:
-
-      * Parallel installability with 0.10.x series
-      * See release notes for more details
diff --git a/RELEASE b/RELEASE
@@ -1,5 +1,5 @@
 
-Release notes for GStreamer Base Plugins 1.0.2
+Release notes for GStreamer Base Plugins 1.1.1
 
 
 The GStreamer team is proud to announce a new bug-fix release
@@ -60,33 +60,96 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
 
 Features of this release
 
-      * Parallel installability with 0.10.x series
-      * alsa: fix probing of supported formats, and advertise non-native-endianness formats as fallback
-      * audiobasesink: properly handle GAP events (fixing some isses with e.g. certain DVD menus)
-      * audioconvert: try harder to not convert or to preserve input format precision
-      * audiodecoder: leak fixes and refcounting fixes
-      * audioresample: re-enable the SSE/SSE2 code paths for better performance
-      * riff: fix paletted RGB formats and msvideo mapping
-      * rtsp: make formatting and parsing of range floating-point values locale-independent
-      * playbin: streamsynchronizer fixes, esp. for handling corner-cases near EOS
-      * tcpserver{sink,src}: add 'current-port' property and signal actually used port
-      * videoconvert: fix handling of paletted RGB formats
-      * videodecoder: don't leak message strings when error is not fatal
-      * videodecoder: finetune missing timestamp estimating
-      * videotestsrc: add palette for paletted RGB formats
-      * vorbistag: fix writing of image tags into vorbis comments
 
 Bugs fixed in this release
 
-      * 580093 : tcpserversink,src: add 'current-port' property and signal actually used port when port=0 was set
-      * 680904 : alsasink: no supported formats detected with using USB sound card on big-endian system
-      * 683098 : videodecoder: log failure message if acquire_buffer failed
-      * 684411 : rtsp: range in SDP formatted according to locale
-      * 685273 : Pre-rolling on GAP events doesn't work properly for audio sinks
-      * 685711 : audio, video: update docs for virtual functions that require chaining up
-      * 685938 : [decodebin] Issues with group switching algorithm
-      * 686081 : adder: all unit tests crash now after collectpads changes
-      * 686298 : Cannot decode some AVI files with Microsoft Video 1
+      * 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
+      * 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
+      * 697112 : GLTextureUploadMeta: No support for multi-texture formats
+      * 634407 : decodebin should expose pads in a deterministic order
+      * 636753 : pbutils function to map (container) caps to filename extension
+      * 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
+      * 663350 : theoraenc: do not reset the encoder when we need a keyframe
+      * 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
+      * 676884 : audiotestsrc: segment one sample too short due to rounding errors
+      * 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
+      * 679456 : videodecoder: fix compiler optimization hint macro usage
+      * 681719 : audiovisualizer does not handle VideoMeta
+      * 685637 : [audioresample] Performance improvements & ARM NEON support
+      * 687146 : rtpbasedepay: remove unused variable
+      * 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
+      * 687466 : audiobasesink: use the same type as the internal type to return it
+      * 687472 : video-blend: fix memory leak
+      * 687817 : textoverlay: support shaded background drawing for all formats
+      * 689326 : multifdsink: document that adding fd in NULL is not allowed
+      * 689845 : Encodebin API to handle multiple streams lacking
+      * 690240 : encodebin: remove test of encoder name vs preset name
+      * 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
+      * 690994 : videodecoder: Allow parse function to not use all data on the adapter
+      * 691072 : decodebin: Doesn't expose pads if no data is received before EOS
+      * 692358 : appsrc deadlock setting the pipeline to NULL state
+      * 692613 : tests: reduce number of wake-ups in test applications
+      * 692930 : avidemux: add raw 8-bit monochrome
+      * 693302 : decodebin: g_mutex_new is deprecated
+      * 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
+      * 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
+      * 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
+      * 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
+      * 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
+      * 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
+      * 694443 : libgstaudio: add support for AAC pass-through
+      * 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
+      * 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
+      * 695276 : libsabi test needs an update for i386
+      * 695540 : riff: support raw avi with negative height
+      * 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
+      * 695660 : appsink: update the emit-signal description
+      * 695832 : audio: a print causes a floating point exception
+      * 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
+      * 696411 : audiotestsrc: incorrect data size in last buffer
+      * 696550 : riff: add " note " tag
+      * 696598 : decodebin pads no longer match order in file
+      * 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
+      * 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
+      * 696916 : videofilter doesn't add caps in pool config
+      * 697628 : ximagesink: Compile error without HAVE_XSHM
+      * 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
+      * 697665 : Add format=WMV3 for WMV 3 video
+      * 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
+      * 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
+      * 697808 : sdp: add boxed type for GstSDPMessage
+      * 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
+      * 698410 : Adder: Can not send flush_start and flush_stop in a row
+      * 698558 : sdp: make it possible to modify session/media attributes
+      * 698712 : playbin: autoplug video decoder and sink based on caps features
+      * 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
+      * 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
+      * 699124 : vorbisdec: crash on shutdown in webkit unit test
+      * 699187 : videorate: ends up outputting buffers with incorrect duration
+      * 699470 : dmabuf: handle mmap failure
+      * 699563 : dmabuf: fix formating
+      * 699565 : dmabuf: fix memory initialization
+      * 699566 : dmabuf: don't touch the GstMemory size
+      * 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
+      * 699792 : oggmux: Never emitting EOS in GES
+      * 699894 : videoencoder: Caps event sent before stream-start
+      * 699960 : videodecoder: Reordering sticky events
+      * 699971 : oggmux: Sends a segment event before sending a caps event.
+      * 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
+      * 700222 : rtpbasepayload: Need to delay segments event after caps event
+      * 700259 : audio: fix buffer overflow for channels > 64
+      * 700272 : playback: Use subset checks instead of intersections
+      * 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
+      * 700377 : video: add NV16 pixel format support
+      * 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
+      * 700411 : dmabuf: Make sure that memory is unmapped before releasing it
+      * 700413 : ximagesink: add alpha mask support
+      * 700427 : dmabuf: set the initial memory size to the full size
+      * 701202 : playsink: Badly initialized contrast/brightness
+      * 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
+      * 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
+      * 589242 : videoconvert: need special handling for interlaced I420
+      * 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
 
 ==== Download ====
 
@@ -123,14 +186,63 @@ subscribe to the gstreamer-devel list.
 
 Contributors to this release
 
-      * Alexandre Relange
+      * Akihiro Tsukada
+      * Alessandro Decina
+      * Alexander Schrab
       * Andoni Morales Alastruey
+      * Anton Belka
+      * Arnaud Vrac
+      * B.Prathibha
+      * Benjamin Gaignard
+      * Brendan Long
       * Carlos Rafael Giani
-      * David Corvoysier
+      * Christian Fredrik Kalager Schaller
+      * Daniel Drake
+      * David Schleef
+      * David Svensson Fors
+      * Dirk Van Haerenborgh
+      * Edward Hervey
+      * Emanuele Aina
+      * Evan Nemerson
+      * Greg Rutz
+      * Jan Schmidt
+      * Jan Schole
+      * Jihyun Cho
+      * Jonas Holmberg
+      * Jonathan Liu
+      * Jose Antonio Santos Cadenas
       * Josep Torra
-      * Mark Nauwelaerts
+      * Julien Moutte
+      * Marc Leeman
+      * Martin Pitt
+      * Matej Knopp
+      * Mathieu Duponchelle
+      * Matthew Waters
+      * Michael Olbrich
+      * Miguel Angel Cabrera Moya
+      * Nicola Murino
+      * Nicolas Dufresne
+      * Ognyan Tonchev
+      * Olivier Crête
+      * Patricia Muscalu
+      * Paul HENRYS
+      * Pete Beardmore
+      * Philippe Normand
+      * Rasmus Rohde
+      * Rico Tzschichholz
       * Sebastian Dröge
+      * Sebastian Rasmussen
+      * Simon Berg
+      * Sreerenj Balachandran
+      * Stefan Sauer
       * Thiago Santos
+      * Thibault Saunier
+      * Thijs Vermeir
+      * Thomas Scheuermann
       * Tim-Philipp Müller
+      * Tom Greenwood
+      * Vincent Penquerc'h
+      * Víctor Manuel Jáquez Leal
       * Wim Taymans
+      * yanghuolin
 
diff --git a/common b/common
diff --git a/configure.ac b/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, git and prerelease does -Werror too
 dnl use a three digit version number for releases, and four for git/prerelease
-AC_INIT([GStreamer Base Plug-ins],[1.1.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
+AC_INIT([GStreamer Base Plug-ins],[1.1.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
 
 AG_GST_INIT
 
@@ -56,10 +56,10 @@ dnl      1.2.5 => 205
 dnl      1.10.9 (who knows) => 1009
 dnl
 dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 100, 0, 100)
+AS_LIBTOOL(GST, 101, 0, 101)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.1.0
+GST_REQ=1.1.1
 
 dnl *** autotools stuff ****
 

diff --git a/docs/plugins/gst-plugins-base-plugins.args b/docs/plugins/gst-plugins-base-plugins.args
@@ -1368,6 +1368,36 @@
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstPlayBin::audio-stream-combiner</NAME>
+<TYPE>GstElement*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Audio stream combiner</NICK>
+<BLURB>Current audio stream combiner (NULL = input-selector).</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPlayBin::text-stream-combiner</NAME>
+<TYPE>GstElement*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Text stream combiner</NICK>
+<BLURB>Current text stream combiner (NULL = input-selector).</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPlayBin::video-stream-combiner</NAME>
+<TYPE>GstElement*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Video stream combiner</NICK>
+<BLURB>Current video stream combiner (NULL = input-selector).</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
 <ARG>
 <NAME>GstAudiorate::add</NAME>
 <TYPE>guint64</TYPE>
@@ -1975,7 +2005,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Add Borders</NICK>
 <BLURB>Add black borders if necessary to keep the display aspect ratio.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -1991,7 +2021,7 @@
 <ARG>
 <NAME>GstVideoScale::envelope</NAME>
 <TYPE>gdouble</TYPE>
-<RANGE>[0,5]</RANGE>
+<RANGE>[1,5]</RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Envelope</NICK>
 <BLURB>Size of filter envelope.</BLURB>
@@ -2011,7 +2041,7 @@
 <ARG>
 <NAME>GstVideoScale::sharpness</NAME>
 <TYPE>gdouble</TYPE>
-<RANGE>[0,2]</RANGE>
+<RANGE>[0.5,1.5]</RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Sharpness</NICK>
 <BLURB>Sharpness of filter.</BLURB>
@@ -3594,7 +3624,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Emit signals</NICK>
-<BLURB>Emit new-preroll, new-buffer and new-buffer-list signals.</BLURB>
+<BLURB>Emit new-preroll and new-sample signals.</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 

diff --git a/docs/plugins/gst-plugins-base-plugins.hierarchy b/docs/plugins/gst-plugins-base-plugins.hierarchy
@@ -2,7 +2,7 @@ GObject
   GInitiallyUnowned
     GstObject
       GstAllocator
-        GstDefaultAllocator
+        GstAllocatorSysmem
       GstAudioRingBuffer
         GstAudioSinkRingBuffer
         GstAudioSrcRingBuffer
@@ -20,7 +20,7 @@ GObject
         GstAudioEncoder
           GstVorbisEnc
         GstAudioRate
-        GstAudioVisualizer-BaseExtVisual
+        GstAudioVisualizer-BaseExtLibvisual
           GstVisual
             GstVisualbumpscope
             GstVisualcorona
@@ -116,6 +116,8 @@ GObject
   GInputStream
   GOutputStream
   GSocket
+  GTypeModule
+    PangoModule
   GstColorBalanceChannel
   GstEncodingProfile
   PangoContext

diff --git a/docs/plugins/gst-plugins-base-plugins.interfaces b/docs/plugins/gst-plugins-base-plugins.interfaces
@@ -1,4 +1,6 @@
 GSocket GInitable
+GTypeModule GTypePlugin
+GstAdder GstChildProxy
 GstAppSink GstURIHandler
 GstAppSrc GstURIHandler
 GstAudioCdSrc GstURIHandler
@@ -23,3 +25,4 @@ GstVorbisTag GstTagSetter
 GstXImageSink GstNavigation GstVideoOverlay
 GstXvImageSink GstNavigation GstVideoOverlay GstColorBalance
 PangoCairoFcFontMap PangoCairoFontMap
+PangoModule GTypePlugin