2 Release notes for GStreamer Base Plugins 1.1.1
5 The GStreamer team is proud to announce a new bug-fix release
6 in the 1.x stable series of the
7 core of the GStreamer streaming media framework.
10 The 1.x series is a stable series targeted at end users.
11 It is not API or ABI compatible with the stable 0.10.x series.
12 It is, however, parallel installable with the 0.10.x series and
13 will not affect an existing 0.10.x installation.
17 This module contains a set of reference plugins, base classes for other
18 plugins, and helper libraries. It also includes essential elements such
19 as audio and video format converters, and higher-level components like playbin,
20 decodebin, encodebin, and discoverer.
22 This module is kept up-to-date together with the core developments. Element
23 writers should look at the elements in this module as a reference for
26 This module contains elements for, among others:
28 device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia
30 codecs: vorbis, theora
31 text: textoverlay, subparse
32 sources: audiotestsrc, videotestsrc, giosrc
35 audio processing: audioconvert, adder, audiorate, audioresample, volume
36 visualisation: libvisual
37 video processing: videoconvert, videoscale
38 high-level components: playbin, uridecodebin, decodebin, encodebin, discoverer
39 libraries: app, audio, fft, pbutils, riff, rtp, rtsp, sdp, tag, video
42 Other modules containing plugins are:
46 contains a set of well-supported plugins under our preferred license
48 contains a set of well-supported plugins, but might pose problems for
51 contains a set of less supported plugins that haven't passed the
52 rigorous quality testing we expect, or are still missing documentation
55 contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
61 Features of this release
64 Bugs fixed in this release
66 * 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
67 * 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
68 * 697112 : GLTextureUploadMeta: No support for multi-texture formats
69 * 634407 : decodebin should expose pads in a deterministic order
70 * 636753 : pbutils function to map (container) caps to filename extension
71 * 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
72 * 663350 : theoraenc: do not reset the encoder when we need a keyframe
73 * 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
74 * 676884 : audiotestsrc: segment one sample too short due to rounding errors
75 * 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
76 * 679456 : videodecoder: fix compiler optimization hint macro usage
77 * 681719 : audiovisualizer does not handle VideoMeta
78 * 685637 : [audioresample] Performance improvements & ARM NEON support
79 * 687146 : rtpbasedepay: remove unused variable
80 * 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
81 * 687466 : audiobasesink: use the same type as the internal type to return it
82 * 687472 : video-blend: fix memory leak
83 * 687817 : textoverlay: support shaded background drawing for all formats
84 * 689326 : multifdsink: document that adding fd in NULL is not allowed
85 * 689845 : Encodebin API to handle multiple streams lacking
86 * 690240 : encodebin: remove test of encoder name vs preset name
87 * 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
88 * 690994 : videodecoder: Allow parse function to not use all data on the adapter
89 * 691072 : decodebin: Doesn't expose pads if no data is received before EOS
90 * 692358 : appsrc deadlock setting the pipeline to NULL state
91 * 692613 : tests: reduce number of wake-ups in test applications
92 * 692930 : avidemux: add raw 8-bit monochrome
93 * 693302 : decodebin: g_mutex_new is deprecated
94 * 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
95 * 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
96 * 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
97 * 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
98 * 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
99 * 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
100 * 694443 : libgstaudio: add support for AAC pass-through
101 * 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
102 * 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
103 * 695276 : libsabi test needs an update for i386
104 * 695540 : riff: support raw avi with negative height
105 * 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
106 * 695660 : appsink: update the emit-signal description
107 * 695832 : audio: a print causes a floating point exception
108 * 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
109 * 696411 : audiotestsrc: incorrect data size in last buffer
110 * 696550 : riff: add " note " tag
111 * 696598 : decodebin pads no longer match order in file
112 * 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
113 * 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
114 * 696916 : videofilter doesn't add caps in pool config
115 * 697628 : ximagesink: Compile error without HAVE_XSHM
116 * 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
117 * 697665 : Add format=WMV3 for WMV 3 video
118 * 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
119 * 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
120 * 697808 : sdp: add boxed type for GstSDPMessage
121 * 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
122 * 698410 : Adder: Can not send flush_start and flush_stop in a row
123 * 698558 : sdp: make it possible to modify session/media attributes
124 * 698712 : playbin: autoplug video decoder and sink based on caps features
125 * 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
126 * 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
127 * 699124 : vorbisdec: crash on shutdown in webkit unit test
128 * 699187 : videorate: ends up outputting buffers with incorrect duration
129 * 699470 : dmabuf: handle mmap failure
130 * 699563 : dmabuf: fix formating
131 * 699565 : dmabuf: fix memory initialization
132 * 699566 : dmabuf: don't touch the GstMemory size
133 * 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
134 * 699792 : oggmux: Never emitting EOS in GES
135 * 699894 : videoencoder: Caps event sent before stream-start
136 * 699960 : videodecoder: Reordering sticky events
137 * 699971 : oggmux: Sends a segment event before sending a caps event.
138 * 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
139 * 700222 : rtpbasepayload: Need to delay segments event after caps event
140 * 700259 : audio: fix buffer overflow for channels > 64
141 * 700272 : playback: Use subset checks instead of intersections
142 * 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
143 * 700377 : video: add NV16 pixel format support
144 * 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
145 * 700411 : dmabuf: Make sure that memory is unmapped before releasing it
146 * 700413 : ximagesink: add alpha mask support
147 * 700427 : dmabuf: set the initial memory size to the full size
148 * 701202 : playsink: Badly initialized contrast/brightness
149 * 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
150 * 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
151 * 589242 : videoconvert: need special handling for interlaced I420
152 * 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
156 You can find source releases of gst-plugins-base in the download
157 directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/
159 The git repository and details how to clone it can be found at
160 http://cgit.freedesktop.org/gstreamer/gst-plugins-base/
164 The project's website is http://gstreamer.freedesktop.org/
166 ==== Support and Bugs ====
168 We use GNOME's bugzilla for bug reports and feature requests:
169 http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
171 Please submit patches via bugzilla as well.
173 For help and support, please subscribe to and send questions to the
174 gstreamer-devel mailing list (see below for details).
176 There is also a #gstreamer IRC channel on the Freenode IRC network.
180 GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
181 from there (see link above).
183 Interested developers of the core library, plugins, and applications should
184 subscribe to the gstreamer-devel list.
187 Contributors to this release
192 * Andoni Morales Alastruey
198 * Carlos Rafael Giani
199 * Christian Fredrik Kalager Schaller
202 * David Svensson Fors
203 * Dirk Van Haerenborgh
213 * Jose Antonio Santos Cadenas
219 * Mathieu Duponchelle
222 * Miguel Angel Cabrera Moya
234 * Sebastian Rasmussen
236 * Sreerenj Balachandran
245 * Víctor Manuel Jáquez Leal