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authorSebastian Dröge <sebastian.droege@collabora.co.uk>2013-06-05 17:58:51 +0200
committerSebastian Dröge <sebastian.droege@collabora.co.uk>2013-06-05 18:31:27 +0200
commitff5d3313d4a8169e26df1dc3391fc229de0b5a89 (patch)
tree7bd3e583df08a5d8fa8ac8bf2ee7b4210491d98d /RELEASE
parent597856bc8c06225d42b94aab09f61c5b114a4ceb (diff)
downloadgstreamer-plugins-base-ff5d3313d4a8169e26df1dc3391fc229de0b5a89.tar.gz
Release 1.1.11.1.1
Diffstat (limited to 'RELEASE')
-rw-r--r--RELEASE168
1 files changed, 140 insertions, 28 deletions
diff --git a/RELEASE b/RELEASE
index 3372b82f3..02b75810e 100644
--- 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
  \ No newline at end of file