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238 lines
10 KiB
Text
238 lines
10 KiB
Text
Release notes for GStreamer Base Plugins 1.3.1
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The GStreamer team is pleased to announce the first release of the unstable
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1.3 release series. The 1.3 release series is adding new features on top of
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the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
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series of the GStreamer multimedia framework. The unstable 1.3 release series
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will lead to the stable 1.4 release series in the next weeks, and newly added
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API can still change until that point.
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Binaries for Android, iOS, Mac OS X and Windows will be provided separately
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during the unstable 1.3 release series.
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The versioning scheme that is used in general is that 1.x.y is API and
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ABI backwards compatible with previous 1.x.y releases. If x is an even
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number it is a stable release series and all releases in this series
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will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
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x is odd it is a development release series that will lead to the next
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stable release series 1.x+1 and contains new features and bigger
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changes. During the development release series, new API can still
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change.
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This module contains a set of reference plugins, base classes for other
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plugins, and helper libraries. It also includes essential elements such
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as audio and video format converters, and higher-level components like playbin,
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decodebin, encodebin, and discoverer.
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This module is kept up-to-date together with the core developments. Element
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writers should look at the elements in this module as a reference for
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their development.
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This module contains elements for, among others:
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device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia
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containers: ogg
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codecs: vorbis, theora
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text: textoverlay, subparse
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sources: audiotestsrc, videotestsrc, giosrc
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network: tcp
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typefind functions
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audio processing: audioconvert, adder, audiorate, audioresample, volume
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visualisation: libvisual
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video processing: videoconvert, videoscale
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high-level components: playbin, uridecodebin, decodebin, encodebin, discoverer
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libraries: app, audio, fft, pbutils, riff, rtp, rtsp, sdp, tag, video
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Other modules containing plugins are:
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gst-plugins-good
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contains a set of well-supported plugins under our preferred license
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gst-plugins-ugly
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contains a set of well-supported plugins, but might pose problems for
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distributors
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gst-plugins-bad
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contains a set of less supported plugins that haven't passed the
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rigorous quality testing we expect, or are still missing documentation
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and/or unit tests
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gst-libav
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contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
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Bugs fixed in this release
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* 684030 : typefinding: mp4 with video and dts ES detected as DTS audio
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* 725078 : audiobasesink: clip start samples to match clipped timestamp from skew algorithm
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* 708633 : adder: Should not take channel mask in consideration when in mono or stereo
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* 540941 : v4l2: RGB32 should be mapped to xRGB instead of RGBx
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* 646577 : rtppayload: Make RTP time information accessible
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* 670690 : audioresample: missing configure checks for SSE / SSE2
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* 678402 : Device discovery/listing replacement for GstPropertyProbe
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* 678590 : subparse: Add support for LRC subtitles
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* 679031 : playbin/playsink: Add support for audio and video filters
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* 687183 : videodecoder: Allow to negotiate a buffer pool before output format is known
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* 702230 : audioringbuffer: Don't access timestamps array if not acquired
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* 707361 : video: Add support for 64x32 tiled NV12 color format
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* 707636 : dashdemux: offline playback not buffering correctly
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* 708680 : typefind: Add typefind function for H265
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* 708921 : pbutils: Add codec-utility functions to support h265
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* 708991 : audiocdsrc: invalid musicbrainz discids because of trailing data tracks
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* 709588 : encodebin: Handle changes in encoding_profile::restriction during playback
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* 709646 : videotestsrc: Could implement duration query when num-buffers is set
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* 709755 : alsa: add channel map API support
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* 709814 : [examples/overlay] avoid to unref sink if not found. Also fix logic to find a sink in one of the example.
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* 709858 : theoraenc: Do nothing when flushing the encoder when no caps were set
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* 710760 : videoconvert: remove unneeded guint comparison
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* 711094 : videodecoder: improve max-error handling
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* 711258 : sdp: fix duplicate 'const' declaration warnings
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* 712798 : videometa: add GstVideoGLTextureUploadMeta buffer pool option
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* 719383 : rtpbasepayload: Perfect timestamps confusingly explained
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* 719415 : rtpbasepayload: Expose running time of last processed buffer
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* 719850 : convertframe: remove trivial memory leak
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* 719890 : videodecoder: Add API to get the currently pending, parsed frame size
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* 720103 : videodecoder: Introduce sink_query/src_query
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* 720124 : tests/examples/overlay/qt-videooverlay.cpp has incorrect include from Qt
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* 720162 : tests: Add test for rtpbasepayload/-depayload
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* 720205 : playback: add video/x-raw(ANY) to default raw caps
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* 720215 : sdp: parse encryption key field
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* 720219 : rtsptransport: allow getting mime type by profile
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* 720389 : videodecoder: should release buffer pool sooner
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* 720810 : audio/video: Initialize all {audio|video}info fields
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* 720999 : Missing annotation for GstColorBalance interface
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* 721103 : test-effect-switch errors out with not-negotiated after a while
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* 721701 : videoconvert: I420 to BGRA conversion is slower than in 0.10
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* 721953 : pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
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* 722330 : streamsplitter: negotiation problems with parsers
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* 722491 : playbin: remove duplicate assignment
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* 722682 : oggmux: problems with vp8 stream
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* 723096 : decodebin: Make it possible to register multiple handlers to decodebin's autoplug-select signal
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* 723271 : videotestsrc: fix a warning if downstream does not propose a buffer pool
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* 723328 : gstrtpbase(|de)payload: add more unit tests and fix bugs
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* 723492 : gst-plugins-base: Do not build check tests for disabled plugins
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* 723507 : jsseek: Add missing HAVE_X check
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* 724393 : rtspconnection: allow specifying an anchor certificate database
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* 724509 : audioconvert: outputs silence when converting certain mono caps to certain other mono caps
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* 724828 : playbin: improve autoplug_query_caps return
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* 724893 : playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
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* 725034 : all plugin sets but -base don't install gtk-doc docs without '--enable-gtk-doc'
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* 725206 : rtspconnection: Missing include file
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* 725479 : gst-plugins-base: Ignore gcov intermediate files
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* 725521 : docs: Fix argument and annotation typos, add missing annotations and remove duplicate section
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* 725658 : Removing some GnomeVFS left bits
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* 725837 : pango: textoverlay: lot of warnings in debug log with framerate=0/1
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* 725878 : rtspconnection: headers in GET response not configurable for tunnels
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* 725898 : Lose data when producing data faster than sendt during tunneling rtps/rtp(TCP)
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* 726433 : rtspconnection: setsockopt() argument 4 is not properly casted for W32
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* 726641 : rtspconnection: connection_poll() not working correctly
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* 727498 : videodecoder: deactivates downstream bufferpool
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* 728772 : rtspconnection: stuck in teardown
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* 728845 : gst-play: add option to supply input media-files from a playlist file
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* 728907 : rtspconnection: add more tests
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* 729114 : audiodecoder: default caps nego will manually fixate non-mutable caps
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* 729117 : rtpbuffer: fix memory leak when gst_rtp_buffer_map fails
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* 729195 : videotestsrc: undefined behaviour in left-shift
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* 729321 : playbin/subtitleoverlay: Deadlock when changing subtitle track while PAUSED
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* 704933 : uridecodebin: allow progressive buffering with more media types
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==== Download ====
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You can find source releases of gst-plugins-base in the download
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directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/
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The git repository and details how to clone it can be found at
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http://cgit.freedesktop.org/gstreamer/gst-plugins-base/
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==== Homepage ====
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The project's website is http://gstreamer.freedesktop.org/
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==== Support and Bugs ====
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We use GNOME's bugzilla for bug reports and feature requests:
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http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
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Please submit patches via bugzilla as well.
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For help and support, please subscribe to and send questions to the
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gstreamer-devel mailing list (see below for details).
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There is also a #gstreamer IRC channel on the Freenode IRC network.
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==== Developers ====
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GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
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from there (see link above).
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Interested developers of the core library, plugins, and applications should
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subscribe to the gstreamer-devel list.
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Contributors to this release
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* Adrien Schwartzentruber
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* Aleix Conchillo Flaque
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* Aleix Conchillo Flaqué
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* Alessandro Decina
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* Andres Gomez
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* Antoine Jacoutot
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* Antonio Ospite
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* Arun Raghavan
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* Bastien Nocera
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* Christian Fredrik Kalager Schaller
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* David Svensson Fors
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* Edward Hervey
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* Eric Trousset
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* George Kiagiadakis
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* Göran Jönsson
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* Haakon Sporsheim
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* Hans Månsson
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* Holger Kaelberer
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* Jan Schmidt
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* Jihyun Cho
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* Johannes Dewender
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* John Bassett
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* Josep Torra
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* Julien Isorce
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* Justin Joy
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* Lionel Landwerlin
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* Luis de Bethencourt
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* Mark Nauwelaerts
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* Matej Knopp
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* Mathieu Duponchelle
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* MathieuDuponchelle
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* Matthew Waters
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* Matthieu Bouron
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* Nicola Murino
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* Nicolas Dufresne
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* Ognyan Tonchev
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* Olivier Crête
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* Rafał Mużyło
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* Ravi Kiran K N
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* Reynaldo H. Verdejo Pinochet
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* Sebastian Dröge
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* Sebastian Rasmussen
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* Sjoerd Simons
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* Sreerenj Balachandran
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* Stefan Sauer
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* Stephan Sundermann
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* Stian Selnes
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* Stéphane Cerveau
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* Takashi Iwai
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* Thiago Santos
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* Thibault Saunier
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* Tim-Philipp Müller
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* Todd Agulnick
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* Tom Greenwood
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* Vincent Penquerc'h
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* William Grant
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* Wim Taymans
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* Wonchul Lee
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* Руслан Ижбулатов
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