+This is GStreamer gst-plugins-good 1.16.0.
-Release notes for GStreamer Good Plug-ins 0.10.29 "Soft Cheese Enthusiast"
-
-
-
-The GStreamer team is proud to announce a new release
-in the 0.10.x stable series of the
-GStreamer Good Plug-ins.
-
-
-The 0.10.x series is a stable series targeted at end users.
-
-
-
-"Such ingratitude. After all the times I've saved your life."
-
-
-A collection of plug-ins you'd want to have right next to you on the
-battlefield. Shooting sharp and making no mistakes, these plug-ins have it
-all: good looks, good code, and good licensing. Documented and dressed up
-in tests. If you're looking for a role model to base your own plug-in on,
-here it is.
-
-
-If you find a plot hole or a badly lip-synced line of code in them,
-let us know - it is a matter of honour for us to ensure Blondie doesn't look
-like he's been walking 100 miles through the desert without water.
-
-
-This module contains a set of plug-ins that we consider to have good quality
- code, correct functionality, our preferred license (LGPL for the plug-in
- code, LGPL or LGPL-compatible for the supporting library).
-We believe distributors can safely ship these plug-ins.
-People writing elements should base their code on these elements.
-
-
-Other modules containing plug-ins are:
-
-
-gst-plugins-base
-contains a basic set of well-supported plug-ins
-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
-
- * audioparser: new amrparse, aacparse, ac3parse, flacparse, mpegaudioparse, dcaparse elements
- * audiowsincband: Add new windowing functions: gaussian, cos and hann
- * audiowsincband: Fix range of kernel elements (lim -> lim-1)
- * audiowsinclimimt: Add new windows to high/low-pass filters: gaussian, cosine, hann
- * audiowsinclimit: Fix range of kernel elements (lim -> lim-1) in high/low-pass filters
- * avidemux: also add the frame-type for the stream index.
- * avidemux, flvdemux: mark delta-units in the index
- * avidemux: stream->current_total is accumulated byte size and not time
- * avimux: add stream-format field to h264 pad template caps
- * avimux: rework _request_new_pad to handle explict req-pad-names
- * avimux: use running time for synchronization
- * cairooverlay: Add generic Cairo overlay video element.
- * debugutils: remove bitrotten negotiation element
- * deinterlace: add support for NV12 and NV21 formats; fix greedyl method
- * dvdemux: first try if upstream handles TIME seeks before handling them here and other event handling fixes
- * flacdec: fix issues with large metadata blocks when streaming unframed flac
- * flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE
- * flacenc: Don't store image tags inside the vorbiscomments and the flac metadata
- * flvdemux: add width, height and framerate to caps when present on onMetaData
- * flvdemux: Do not build an index if upstream is not seekable
- * flvdemux: fix deadlock on setting index on flvdemux
- * flvmux: don't overwrite metadata tag with duration in streaming mode
- * flvmux: don't set duration for live stream
- * flvmux: use running time for synchronization
- * flv: specify stream-format for h264 in the pad template caps
- * icydemux: fix tag list handling issues that might have caused crashes
- * j2kpay: skip EPH packets
- * jitterbuffer: also estimate eos if very near eos
- * jitterbuffer: avoid trying to buffer more than is available
- * jitterbuffer: handle position query
- * matroskademux: better calculation of output framerate
- * matroskademux: properly resume cluster scanning
- * matroskademux: pull mode should always report seekable
- * matroskademux: set stream-format=byte-stream on h264 caps if there's no codec data
- * matroskademux: store cluster positions provided by SeekHead
- * matroskamux: add support for A-Law and µ-Law
- * matroskamux: avoid building index when streamable
- * matroskamux: use running time for stream synchronization
- * matroskamux: add stream-format field to h264 pad template caps
- * matroska: Use ARTIST Matroska tag instead of AUTHOR for GST_TAG_ARTIST
- * matroskaparse: new element
- * monoscope: stability (off-by-one) and memory leak fixes
- * pngdec: handle 16-bit-per-channel images
- * pulsesink: also uncork during EOS waiting (and after EOS is rendered)
- * pulsesink: fix deadlock if connecting to PA fails
- * pulsesink: release pa_shared_resource_mutex before pa_threaded_mainloop_wait
- * qtdemux: Adds more h264 fields to its caps
- * qtdemux: Add support for 2Vuy and r210
- * qtdemux: don't error out when there's a problem parsing non-vital headers
- * qtdemux: avoid skipping exposing a stream following a removed stream
- * qtdemux: Check for invalid (empty) classification info entity strings
- * qtdemux: extract MusicBrainz tags
- * qtdemux: mind rounding issues when converting from global time to mov time
- * qtdemux: propagate error during expose_streams
- * qtdemux: support some more mpeg-4 fourcc variants
- * qtdemux: take configured start time into account
- * isomp4: move mp4mux/3gppmux/qtmux from -bad to -good, rename qtdemux plugin to isomp4
- * rtpbin: Don't try to request the same request pad twice
- * rtpbin: fix setting the SDES property
- * rtpbin: Get and use the NTP time when receiving RTCP
- * rtpmanager: ignore a BYE if it is sent with our internal SSRC
- * rtpptdemux: Tag upstream custom events with payload type
- * rtpsession: add action signal to request early RTCP
- * rtpsession: add "rtcp-min-interval" property for minimum interval between Regular RTCP messages
- * rtpsession: Don't relay more than one PLI request per RTT
- * rtpsession: Emit "on-ssrc-validated" when validating by RTCP
- * rtpsession: Emit signal on incoming RTCP feedback packet
- * rtpsession: Emit signal when sending a compound RTCP packet
- * rtpsession: Implement sending PLI packets in response to GstForceKeyUnit
- * rtpsession: Number of active sources should be updated whenever the status of the source changes to active
- * rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI
- * rtpsource: Retain RTCP Feedback packets for a specified amount of time
- * rtpssrcdemux: Tag upstream custom events with SSRC
- * rtpssrcdemux: Unknown SSRC is not fatal
- * rtpspeexpay: Do not transmit samples with GAP flag
- * rtptheoradepay: Request new keyframe on lost packets
- * rtpvrawpay: add support for interlaced video
- * rtspsrc: distribute new base_time to manager children following flush seek
- * rtspsrc: handle * control correctly
- * rtspsrc: improve recovery from failed seek
- * spectrum: miscellaneous optimisations, add multi-channel support
- * speexdec: Always process the number of frames per packet as specified in the header
- * speexdec: get and use streamheader from the caps if possible
- * speexenc: Use speex intern silence detection
- * theorapay: handle 0-sized packets (which are repeat frames)
- * udpsink: warn when packet is too large
- * v4l2: Add PJPG mapping
- * v4l2: fix interlaced set_format configuration
- * v4l2: new v4l2radio element to control analog radio devices
- * videobalance: fix handling of YUV images with 'odd' widths
- * videoflip: add support for YUY2, UVYV and YVYU
- * videoflip: fix invalid memory access for odd resolutions and Y422
- * videomixer2: Add transparent background option for alpha channel formats
- * videomixer: Add transparent background option for alpha channel formats
- * videomixer: Fix argb/rgba overlay orc code
- * wavparse: tune output max buffer size to material
-
-Bugs fixed in this release
-
- * 564122 : Crash in monoscope_update
- * 432612 : [matroskamux] doesn't handle segments correctly
- * 593482 : Spectrum: Multi-Channel support and Stereo to Mono compat report(cross-correlation)
- * 595520 : Implement a generic cairo overlay
- * 622553 : rtpmanager: Implement RFC 4585 (AVPF / early feedback)
- * 636699 : [PLUGIN-MOVE] qtmux: move to -good
- * 639994 : videomixer2: added 'transparent' background option
- * 640118 : v4l2: add element to control radio devices
- * 640163 : rtspsrc: minor leak
- * 640249 : [taginject] Taginject does not allow to change tags after init
- * 640483 : flvdemux: Video's width, height and/or framerate src caps added when present on onMetaData
- * 640542 : matroskamux leaks memory after reset
- * 641330 : icydemux: crash while playing MP3 stream in amarok
- * 641332 : can't connect vorbisenc ! queue ! matroskamux
- * 641400 : [deinterlace] Handle image caps without asserting
- * 641827 : rtptheorapay: doesn't handle 0-size packets
- * 642205 : qtdemux: extract MusicBrainz tags
- * 642337 : [souphttpsrc] Add support for URI queries
- * 642412 : gstrtpbin with ignore-pt tries to use NULL stream- > demux during uninitialization
- * 642691 : deinterlace: Miscellaneous cleanup
- * 642879 : qtmux: add a 'variant' with the bare video/quicktime media type
- * 642961 : NV12 colorspace support for deinterlace plugin
- * 642963 : [dvdemux] time based upstream seek
- * 643087 : pulsesink: deadlock in gst_pulseringbuffer_open_device
- * 643981 : [cairooverlay] example uses gtk/gtk-x11 unnecessarily
- * 644288 : generic/states check fails
- * 644477 : [jack] doesn't build with jack > = 0.120.2
- * 644510 : pulsesink: deadlock when create/connect fails
- * 644669 : gstspeexdec causes 'Conditional jump depends on uninitialised value'
- * 644773 : Add support for Y422 colorspaces in videoflip element
- * 644849 : [speexdec] Remove warning message when it is inappropriate
- * 644875 : [matroskademux] can't read the ARTIST tag in a Matroska file
- * 645858 : [flvdemux] memory leak when demuxing infinite FLV files
- * 645961 : [pulsesink] hangs when going from paused to playing near EOS
- * 646397 : rtpjitterbuffer base_time broken by commit f84b8a69
- * 646474 : rtpspeexpay should drop empty samples
- * 646567 : [matroska] Add alaw/mulaw audio support
- * 646800 : rtspsrc: control attribute on the session and not on the media
- * 646954 : rtpgstpay: declare frag_offset to hold 32 bits
- * 646964 : rtpmanager: ignore a BYE if it is sent with our internal SSRC
- * 646965 : rtpmanager: Number of active sources should be updated whenever the status of the source changes to active
- * 646966 : rtpssrcdemux: Unknown SSRC is not fatal
- * 646967 : rtpsession: make iterate_internal_links MT-safe
- * 646999 : [regression] pulsesink: underruns while playing WMA
- * 647263 : REGRESSION: rtpsession: fix wrongly applied patch
- * 647510 : audiowsinclimit uses the wrong limits for the range of the kernel elements
- * 647511 : add other common windows to low/high-pass filters in audiowsinclimit.c
- * 647659 : mp3parse / mpegaudioparse fails to detect VBRI header in mpeg1 mono and mpeg2 files
- * 647833 : matroskademux: bad at guessing the framerate
- * 647848 : Failure to compile with GCC 4.6.x due to variable unused but set warnings being treated as errors
- * 647919 : qtmux: silently corrupts h264 streams with legacy caps
- * 648004 : [quicktime] Rename plugin library to quicktime too
- * 648160 : Remove half-complete bits of RTCP FIR support
- * 648589 : jpegdec: documentation typo " jpegddec "
- * 649060 : flvmux: overwrites metadata tags with duration in streamable=false mode
- * 649449 : [gppmux] Failure to write location
- * 566769 : [flacdec] crash in push mode with large header packet (image)
- * 644730 : [matroskamux] Should return TRUE in the event function when the event is handled
-
-Download
-
-You can find source releases of gst-plugins-good in the download directory:
-http://gstreamer.freedesktop.org/src/gst-plugins-good/
-
-GStreamer Homepage
-
-More details can be found on the project's website:
-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
-
-Developers
-
-GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there.
-Interested developers of the core library, plug-ins, and applications should
-subscribe to the gstreamer-devel list. If there is sufficient interest we
-will create more lists as necessary.
-
-
-Applications
-
-Contributors to this release
-
- * "Carsten Kroll
- * Alejandro Gonzalez
- * Alessandro Decina
- * Alexey Chernov
- * Alexey Fisher
- * Andoni Morales Alastruey
- * Arun Raghavan
- * Benjamin Otte
- * Christian Fredrik Kalager Schaller
- * Christian Schaller
- * David Hoyt
- * David Schleef
- * Edward Hervey
- * Felipe Contreras
- * Haakon Sporsheim
- * Havard Graff
- * Jan Schmidt
- * Jan Urbanski
- * Jan Urbański
- * Jon Nordby
- * Jordi Burguet-Castell
- * Josep Torra
- * Joshua M. Doe
- * Julien Moutte
- * LRN
- * Lane Brooks
- * Lasse Laukkanen
- * Leonardo Sandoval
- * Luis de Bethencourt
- * Marc-André Lureau
- * Mark Nauwelaerts
- * Michael Smith
- * Ole André Vadla Ravnås
- * Olivier Crête
- * Pascal Buhler
- * Philip Jägenstedt
- * Philippe Normand
- * René Stadler
- * Rob Clark
- * Robert Swain
- * Sebastian Dröge
- * Stefan Kost
- * Thiago Santos
- * Thiago Sousa Santos
- * Thibault Saunier
- * Tim-Philipp Müller
- * Tom Janiszewski
- * Vincent Penquerc'h
- * Wim Taymans
- * Youness Alaoui
- * Zaheer Abbas Merali
-
\ No newline at end of file
+The GStreamer team is thrilled to announce a new major feature release in the
+stable 1.0 API series of your favourite cross-platform multimedia framework!
+
+As always, this release is again packed with new features, bug fixes and
+other improvements.
+
+The 1.16 release series adds new features on top of the 1.14 series and is
+part of the API and ABI-stable 1.x release series of the GStreamer multimedia
+framework.
+
+Full release notes will one day be found at:
+
+ https://gstreamer.freedesktop.org/releases/1.16/
+
+Binaries for Android, iOS, Mac OS X and Windows will usually be provided
+shortly after the release.
+
+This module will not be very useful by itself and should be used in conjunction
+with other GStreamer modules for a complete multimedia experience.
+
+ - gstreamer: provides the core GStreamer libraries and some generic plugins
+
+ - gst-plugins-base: a basic set of well-supported plugins and additional
+ media-specific GStreamer helper libraries for audio,
+ video, rtsp, rtp, tags, OpenGL, etc.
+
+ - gst-plugins-good: a set of well-supported plugins under our preferred
+ license
+
+ - gst-plugins-ugly: a set of well-supported plugins which might pose
+ problems for distributors
+
+ - gst-plugins-bad: a set of plugins of varying quality that have not made
+ their way into one of core/base/good/ugly yet, for one
+ reason or another. Many of these are are production quality
+ elements, but may still be missing documentation or unit
+ tests; others haven't passed the rigorous quality testing
+ we expect yet.
+
+ - gst-libav: a set of codecs plugins based on the ffmpeg library. This is
+ where you can find audio and video decoders and encoders
+ for a wide variety of formats including H.264, AAC, etc.
+
+ - gstreamer-vaapi: hardware-accelerated video decoding and encoding using
+ VA-API on Linux. Primarily for Intel graphics hardware.
+
+ - gst-omx: hardware-accelerated video decoding and encoding, primarily for
+ embedded Linux systems that provide an OpenMax
+ implementation layer such as the Raspberry Pi.
+
+ - gst-rtsp-server: library to serve files or streaming pipelines via RTSP
+
+ - gst-editing-services: library an plugins for non-linear editing
+
+==== Download ====
+
+You can find source releases of gstreamer in the download
+directory: https://gstreamer.freedesktop.org/src/gstreamer/
+
+The git repository and details how to clone it can be found at
+https://cgit.freedesktop.org/gstreamer/gstreamer/
+
+==== Homepage ====
+
+The project's website is https://gstreamer.freedesktop.org/
+
+==== Support and Bugs ====
+
+We have recently moved from GNOME Bugzilla to GitLab on freedesktop.org
+for bug reports and feature requests:
+
+ https://gitlab.freedesktop.org/gstreamer
+
+Please submit patches via GitLab as well, in form of Merge Requests. See
+
+ https://gstreamer.freedesktop.org/documentation/contribute/
+
+for more details.
+
+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 source code repositories can be found on GitLab on freedesktop.org:
+
+ https://gitlab.freedesktop.org/gstreamer
+
+and can also be cloned from there and this is also where you can submit
+Merge Requests or file issues for bugs or feature requests.
+
+Interested developers of the core library, plugins, and applications should
+subscribe to the gstreamer-devel list:
+
+ https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
projects / platform / upstream / gst-plugins-good.git / blobdiff