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248 lines
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Release notes for GStreamer Base Plugins 1.1.1
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The GStreamer team is proud to announce a new bug-fix release
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in the 1.x stable series of the
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core of the GStreamer streaming media framework.
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The 1.x series is a stable series targeted at end users.
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It is not API or ABI compatible with the stable 0.10.x series.
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It is, however, parallel installable with the 0.10.x series and
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will not affect an existing 0.10.x installation.
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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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Features of this release
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Bugs fixed in this release
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* 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
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* 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
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* 697112 : GLTextureUploadMeta: No support for multi-texture formats
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* 634407 : decodebin should expose pads in a deterministic order
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* 636753 : pbutils function to map (container) caps to filename extension
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* 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
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* 663350 : theoraenc: do not reset the encoder when we need a keyframe
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* 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
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* 676884 : audiotestsrc: segment one sample too short due to rounding errors
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* 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
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* 679456 : videodecoder: fix compiler optimization hint macro usage
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* 681719 : audiovisualizer does not handle VideoMeta
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* 685637 : [audioresample] Performance improvements & ARM NEON support
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* 687146 : rtpbasedepay: remove unused variable
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* 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
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* 687466 : audiobasesink: use the same type as the internal type to return it
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* 687472 : video-blend: fix memory leak
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* 687817 : textoverlay: support shaded background drawing for all formats
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* 689326 : multifdsink: document that adding fd in NULL is not allowed
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* 689845 : Encodebin API to handle multiple streams lacking
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* 690240 : encodebin: remove test of encoder name vs preset name
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* 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
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* 690994 : videodecoder: Allow parse function to not use all data on the adapter
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* 691072 : decodebin: Doesn't expose pads if no data is received before EOS
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* 692358 : appsrc deadlock setting the pipeline to NULL state
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* 692613 : tests: reduce number of wake-ups in test applications
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* 692930 : avidemux: add raw 8-bit monochrome
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* 693302 : decodebin: g_mutex_new is deprecated
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* 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
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* 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
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* 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
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* 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
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* 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
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* 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
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* 694443 : libgstaudio: add support for AAC pass-through
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* 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
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* 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
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* 695276 : libsabi test needs an update for i386
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* 695540 : riff: support raw avi with negative height
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* 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
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* 695660 : appsink: update the emit-signal description
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* 695832 : audio: a print causes a floating point exception
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* 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
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* 696411 : audiotestsrc: incorrect data size in last buffer
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* 696550 : riff: add " note " tag
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* 696598 : decodebin pads no longer match order in file
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* 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
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* 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
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* 696916 : videofilter doesn't add caps in pool config
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* 697628 : ximagesink: Compile error without HAVE_XSHM
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* 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
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* 697665 : Add format=WMV3 for WMV 3 video
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* 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
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* 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
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* 697808 : sdp: add boxed type for GstSDPMessage
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* 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
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* 698410 : Adder: Can not send flush_start and flush_stop in a row
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* 698558 : sdp: make it possible to modify session/media attributes
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* 698712 : playbin: autoplug video decoder and sink based on caps features
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* 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
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* 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
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* 699124 : vorbisdec: crash on shutdown in webkit unit test
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* 699187 : videorate: ends up outputting buffers with incorrect duration
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* 699470 : dmabuf: handle mmap failure
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* 699563 : dmabuf: fix formating
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* 699565 : dmabuf: fix memory initialization
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* 699566 : dmabuf: don't touch the GstMemory size
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* 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
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* 699792 : oggmux: Never emitting EOS in GES
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* 699894 : videoencoder: Caps event sent before stream-start
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* 699960 : videodecoder: Reordering sticky events
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* 699971 : oggmux: Sends a segment event before sending a caps event.
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* 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
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* 700222 : rtpbasepayload: Need to delay segments event after caps event
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* 700259 : audio: fix buffer overflow for channels > 64
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* 700272 : playback: Use subset checks instead of intersections
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* 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
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* 700377 : video: add NV16 pixel format support
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* 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
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* 700411 : dmabuf: Make sure that memory is unmapped before releasing it
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* 700413 : ximagesink: add alpha mask support
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* 700427 : dmabuf: set the initial memory size to the full size
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* 701202 : playsink: Badly initialized contrast/brightness
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* 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
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* 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
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* 589242 : videoconvert: need special handling for interlaced I420
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* 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
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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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* Akihiro Tsukada
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* Alessandro Decina
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* Alexander Schrab
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* Andoni Morales Alastruey
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* Anton Belka
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* Arnaud Vrac
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* B.Prathibha
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* Benjamin Gaignard
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* Brendan Long
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* Carlos Rafael Giani
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* Christian Fredrik Kalager Schaller
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* Daniel Drake
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* David Schleef
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* David Svensson Fors
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* Dirk Van Haerenborgh
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* Edward Hervey
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* Emanuele Aina
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* Evan Nemerson
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* Greg Rutz
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* Jan Schmidt
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* Jan Schole
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* Jihyun Cho
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* Jonas Holmberg
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* Jonathan Liu
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* Jose Antonio Santos Cadenas
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* Josep Torra
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* Julien Moutte
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* Marc Leeman
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* Martin Pitt
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* Matej Knopp
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* Mathieu Duponchelle
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* Matthew Waters
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* Michael Olbrich
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* Miguel Angel Cabrera Moya
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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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* Patricia Muscalu
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* Paul HENRYS
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* Pete Beardmore
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* Philippe Normand
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* Rasmus Rohde
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* Rico Tzschichholz
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* Sebastian Dröge
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* Sebastian Rasmussen
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* Simon Berg
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* Sreerenj Balachandran
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* Stefan Sauer
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* Thiago Santos
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* Thibault Saunier
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* Thijs Vermeir
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* Thomas Scheuermann
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* Tim-Philipp Müller
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* Tom Greenwood
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* Vincent Penquerc'h
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* Víctor Manuel Jáquez Leal
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* Wim Taymans
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* yanghuolin
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