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Release notes for
GStreamer Base Plugins 1.7.1
The GStreamer team is pleased to announce the first release of the unstable
1.7 release series. The 1.7 release series is adding new features on top of
the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.7 release series
will lead to the stable 1.8 release series in the next weeks. Any newly added
API can still change until that point.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.7 release series.
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)
Bugs fixed in this release
-
681447
: video overlay composition: fix video blending over transparent frame
-
705579
: Playbin prevents plugins requesting a GstContext to work properly
-
726117
: typefinding: issue in MPEG-TS detection logic for streams with Null Pids
-
726472
: rtpbasepayload: Implement video SDP attributes
-
727970
: videorate: remove dead code
-
730926
: tags: add GST_TAG_PRIVATE_DATA and expose ID3 private frame ( " PRIV " ) data
-
731791
: videometa: add GstVideoAffineTransformationMeta
-
738687
: midi: add alsamidisrc, an ALSA MIDI sequencer source
-
749596
: rtsp-over-http authentication failure
-
751470
: encodebin: Fix special case.
-
752651
: decodebin: segfault on setting to NULL
-
753852
: gstreamer: base: Fix memory leaks when context parse fails.
-
754054
: videorate: remove unnecessary break statement
-
754196
: audiodecoder-test: port to using GstHarness
-
754223
: audioencoder-tests: port to use GstHarness
-
754450
: audiotestsrc: remove frequency and channel number limit
-
755260
: decodebin: Fix a race condition accessing the decode_chain field.
-
755301
: audioconvert: Integer- > Float conversion creates values slightly smaller than -1.0
-
755440
: gst-play: Add keyboard shortcut '0' to seek to beginning
-
755482
: videotestsrc: Force alpha downstream if foreground color contains alpha
-
756804
: playsink: text_sink dynamic reconnection is not working
-
757008
: tests: typefindfunctions: Fix error leak
-
757068
: audio{filter,convert,resample}: Clip input buffers to the segment before handling them
-
757351
: audioconvert: Latest audioconvert outputs noise
-
757480
: Use GST_STIME_FORMAT and GST_STIME_ARGS with GstClockTimeDiff
-
757926
: pbutils:encoding-target: Fix string memory leak
-
757927
: tests:video: Fix overlay rectangle and buffer leak
-
757928
: audio-quantize: Fix dither_buffer memory leak
-
758235
: rtspconnection: add support for parsing custom headers
-
758744
: allocators: Add logging category for GstFdMemory
-
758911
: audiobasesink/src: send latency message on setcaps
-
758922
: rtspconnection should optionally make HTTP requests with abs_path instead of absoluteURI
-
759126
: appsrc: issues with duration query handling
-
759329
: convertframe: Support video crop when convert frame
-
759356
: encodebin: Implement an encoding profile serialization format
-
742875
: [API] new audiovisualizer base class
-
758754
: oggdemux: failing to play an Opus sample file
Download
You can find source releases of gst-plugins-base in the
gst-plugins-base download directory.
The git repository and details how to clone it can be found at
git.freedesktop.org
.
Homepage
The project's website is https://gstreamer.freedesktop.org.
Support and Bugs
We use GNOME's bugzilla for
bug reports and feature requests.
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).
Find us on IRC at #gstreamer.
Developers
Git is hosted on git.freedesktop.org. You can
browse the gst-plugins-base repository.
All code is in Git and can be checked out from there.
Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Contributors to this release
- Andreas Frisch
- Antonio Ospite
- Arnaud Vrac
- Csaba Toth
- Edward Hervey
- Eunhae Choi
- Evan Callaway
- Guillaume Desmottes
- Havard Graff
- Jan Schmidt
- Joan Pau Beltran
- Julien Isorce
- Kazunori Kobayashi
- Koop Mast
- Luis de Bethencourt
- Mathieu Duponchelle
- Matthew Waters
- Michael Olbrich
- Miguel París Díaz
- Nicolas Dufresne
- Nirbheek Chauhan
- Ognyan Tonchev
- Pankaj Darak
- Pavel Bludov
- Perry Hung
- Philippe Normand
- Rajat Verma
- Ravi Kiran K N
- Reynaldo H. Verdejo Pinochet
- Sebastian Dröge
- Sebastian Rasmussen
- Song Bing
- Stefan Sauer
- Stian Selnes
- Thiago Santos
- Thibault Saunier
- Thomas Bluemel
- Tim-Philipp Müller
- Vincent Penquerc'h
- Vineeth T M
- Vineeth TM
- Vivia Nikolaidou
- William Manley
- Wim Taymans
- Xavier Claessens
- eunhae choi
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