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Release notes for GStreamer Base Plug-ins 0.10.15 "No need to argue"

The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Base Plug-ins.

The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series.

This module contains a set of reference plugins, base classes for other plugins, and helper libraries. 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, gnomevfssrc
  • network: tcp
  • typefind
  • audio processing: audioconvert, adder, audiorate, audioscale, volume
  • visualisation: libvisual
  • video processing: ffmpegcolorspace
  • aggregate elements: decodebin, playbin
Other modules containing plug-ins are:
gst-plugins-good
contains a set of well-supported plug-ins under our preferred license
gst-plugins-ugly
contains a set of well-supported plug-ins, but might pose problems for distributors
gst-plugins-bad
contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect

Features of this release

  • RTP/RTSP/RTCP/SDP support improved
  • New FFT support library libgstfft, based on Kiss FFT
  • New formats supported in volume and audiotestsrc
  • Fixes in audiorate and videorate
  • Audio capture fixes
  • Playbin and decodebin fixes
  • New tagdemux base class for ID3/APE style tag readers
  • Fix a nasty crash in the X sinks on shutdown
  • New tags supported
  • Add support for multichannel WAV files.
  • Preserve channel layout information when up/down-mixing.
  • Many bug-fixes and improvements

Bugs fixed in this release

  • 475395 : decodebin2 leaks request-pads
  • 475451 : [decodebin2] leaks ghostpad
  • 378770 : [xvimagesink] race condition in event thread?
  • 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
  • 430677 : [audioconvert] does not preserve channel positions when f...
  • 442654 : [volume] controller bypassed by default
  • 445529 : [volume] support for 24/32-bit audio/x-raw-int
  • 446766 : return code for gst_base_rtp_payload_audio_handle_event()
  • 451970 : Subparse requires HTML parser
  • 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
  • 459334 : [textoverlay] expose pango line alignment property
  • 459585 : [basertpdepayload] api without namespace
  • 460422 : [audiotestsrc] Add support for float and double output
  • 462805 : [alsa] compilation fails with gcc 4.2
  • 462979 : Add 'silent' property to GstTimeOverlay
  • 463215 : [audioconvert] compile errors
  • 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
  • 464666 : [playbin] QT trailer hangs in preroll with decodebin2
  • 464690 : Add connection-speed property to uridecodebin element
  • 465015 : [playbin] Not removed probes causes deadlocks in streamin...
  • 465028 : some warnings with mingw
  • 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
  • 468129 : [basertpaudiopayload] event handler returns the wrong value
  • 468619 : New library gstfft: FFT library for integer and float typ...
  • 470456 : [API] add gst_missing_*_installer_detail_new()
  • 470766 : [ssaparse] line breaks in SSA subtitle parser
  • 471067 : Make the SDP code useable for generating SDP descriptions
  • 471194 : [rtpbuffer] RTP headers are wrong for win32
  • 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
  • 474384 : gstrtsp-enumtypes.c and .h needed for win32
  • 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
  • 475731 : rtspconnection is able to read incomplete messages
  • 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
  • 484989 : memleak, not unrefed caps for gstbasertppayload.c
  • 489010 : Please change default channel order for WAVE_EXT-less .wa...
  • 491722 : [playbin] regression: crash with external subtitles
  • 492098 : [GstFFT] Broken scaling
  • 492114 : Build issues on Windows/MSVC
  • 492306 : compilation errors with MinGW
  • 492813 : Missing symbols in libgstrtp.def
  • 493986 : Build issues on Windows (missing symbols)
  • 494346 : pre-release vs6 patch
  • 496548 : Including malloc.h breaks macos build
  • 496724 : DSW file references non-existent DSP files
  • 464079 : audiotestsrc doesn't respond to conversion queries properly
  • 442065 : floatcast.h includes config.h and might break other apps
  • 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
  • 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
  • 464028 : Move connection-speed from playbin to playbasebin

API changes

  • API additions
    • GstTagDemux base class for simple tag demuxers
    • GstBaseAudioSrc::provide-clock property
    • gst_rtcp_ntp_to_unix()
    • gst_rtcp_unix_to_ntp()
    • gst_rtp_buffer_get_header_len()
    • gst_rtp_buffer_get_extension_data()
    • gst_rtp_buffer_compare_seqnum()
    • gst_rtp_buffer_ext_timestamp()
    • gst_rtcp_packet_sdes_copy_entry()
    • gst_install_plugins_supported()
    • gst_missing_*_installer_detail_new() convenience API
    • gst_rtsp_connection_poll()
    • GstTextOverlay::line-alignment property

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).

There is also a #gstreamer IRC channel on the Freenode IRC network.

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.

Applications

Contributors to this release

  • Stefan Kost
  • Alexander Shopov
  • Damien Lespiau
  • Dan Williams
  • Daniel Díaz
  • David Schleef
  • Davyd Madeley
  • Funda Wang
  • Haakon Sporsheim
  • Ilkka Tuohela
  • Jakub Bogusz
  • Jan Schmidt
  • Jason Kivlighn
  • Jens Granseuer
  • Johan Dahlin
  • Jorge González González
  • Josep Torra Valles
  • Julien MOUTTE
  • Laurent Glayal
  • Michael Smith
  • Mogens Jaeger
  • Ole André Vadla Ravnås
  • Olivier Crete
  • Peter Kjellerstedt
  • Renato Filho
  • René Stadler
  • Sebastian Dröge
  • Sebastien Moutte
  • Stefan Kost
  • Thijs Vermeir
  • Thomas Vander Stichele
  • Tim-Philipp Müller
  • Tommi Myöhänen
  • Vincent Torri
  • Wim Taymans
  • Yang Hong

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