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RELEASE
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Release notes for GStreamer Base Plugins 1.1.1
The GStreamer team is proud to announce a new bug-fix release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
The 1.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.10.x series.
It is, however, parallel installable with the 0.10.x series and
will not affect an existing 0.10.x installation.
This module contains a set of reference plugins, base classes for other
plugins, and helper libraries. It also includes essential elements such
as audio and video format converters, and higher-level components like playbin,
decodebin, encodebin, and discoverer.
This module is kept up-to-date together with the core developments. Element
writers should look at the elements in this module as a reference for
their development.
This module contains elements for, among others:
device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia
containers: ogg
codecs: vorbis, theora
text: textoverlay, subparse
sources: audiotestsrc, videotestsrc, giosrc
network: tcp
typefind functions
audio processing: audioconvert, adder, audiorate, audioresample, volume
visualisation: libvisual
video processing: videoconvert, videoscale
high-level components: playbin, uridecodebin, decodebin, encodebin, discoverer
libraries: app, audio, fft, pbutils, riff, rtp, rtsp, sdp, tag, video
Other modules containing plugins are:
gst-plugins-good
contains a set of well-supported plugins under our preferred license
gst-plugins-ugly
contains a set of well-supported plugins, but might pose problems for
distributors
gst-plugins-bad
contains a set of less supported plugins that haven't passed the
rigorous quality testing we expect, or are still missing documentation
and/or unit tests
gst-libav
contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Features of this release
Bugs fixed in this release
* 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 ====
You can find source releases of gst-plugins-base in the download
directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/
The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-plugins-base/
==== Homepage ====
The project's website is http://gstreamer.freedesktop.org/
==== Support and Bugs ====
We use GNOME's bugzilla for bug reports and feature requests:
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
Please submit patches via bugzilla as well.
For help and support, please subscribe to and send questions to the
gstreamer-devel mailing list (see below for details).
There is also a #gstreamer IRC channel on the Freenode IRC network.
==== Developers ====
GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
from there (see link above).
Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Contributors to this release
* Akihiro Tsukada
* Alessandro Decina
* Alexander Schrab
* Andoni Morales Alastruey
* Anton Belka
* Arnaud Vrac
* B.Prathibha
* Benjamin Gaignard
* Brendan Long
* Carlos Rafael Giani
* 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
* 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