+This is GStreamer gst-plugins-good 1.16.1.
-Release notes for GStreamer Good Plugins 1.5.1
-
-
-The GStreamer team is pleased to announce the first release of the unstable
-1.5 release series. The 1.5 release series is adding new features on top of
-the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
-series of the GStreamer multimedia framework. The unstable 1.5 release series
-will lead to the stable 1.6 release series in the next weeks, and 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 1.5 release series.
-
-
-
-"Such ingratitude. After all the times I've saved your life."
-
-
-A collection of plugins you'd want to have right next to you on the
-battlefield. Shooting sharp and making no mistakes, these plugins 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 plugin 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 plugins that we consider to have good quality
- code, correct functionality, our preferred license (LGPL for the plugin
- code, LGPL or LGPL-compatible for the supporting library).
-We believe distributors can safely ship these plugins.
-People writing elements should base their code on these elements.
-
-
-Other modules containing plugins are:
-
-
-gst-plugins-base
-contains a basic set of well-supported plugins
-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
-
- * 740130 : matroskamux: wrong duration on some files
- * 699382 : v4l2: dmabuf handling is not complete
- * 746747 : rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
- * 741783 : qtmux: crash when trying to mux ALAC
- * 601733 : rtspsrc: Use specific error message when authentication is required
- * 635701 : rtspsrc: seeking is broken
- * 678124 : multifilesink: add support for time based file switching
- * 682770 : v4l2src: should renegotiate
- * 690646 : ximagesrc: Cursor offset with ximagesrc and xid
- * 690719 : jackaudiosink: add new property (port-pattern) to specify which jack ports to autoconnect to
- * 692473 : qtmux: does not store stream specific tags
- * 708808 : qtmux: Error out when downstream is not seekable and no fast-start
- * 711764 : osxaudiosrc: Produces broken audio for any sample rate other than 44100Hz
- * 722567 : wavparse: loops on incorrect wav file
- * 725335 : rtspsrc: Extract the payload type from sdp framesize attribute
- * 726415 : rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute
- * 726416 : rtph263pay/-depay: add framesize SDP attribute
- * 730417 : rtspt: no timestamp from some rtsp source over tcp
- * 731038 : playbin downmixes 5.0 multichannel-audio to stereo
- * 732152 : multiudpsink: use sendmmsg() to send multiple packets to multiple recipients in one go
- * 732866 : udpsink: client add/remove from app blocked while render function is stuck in g_socket_send_message()
- * 732870 : jpegenc: add support for encoding from nv21
- * 733225 : Lockup while using Cheese on 1.3.91
- * 733444 : wavenc: does not support more than 2 channel
- * 733539 : rtph264pay: append profile-level-id parameter to SDP if available
- * 733556 : h264 payloader : append packetization-mode parameter for SDP
- * 733616 : v4l2object: code cleanup
- * 733750 : v4l2object: query minimum required buffers for output
- * 734322 : RTP Jitterbuffer shouldn't force clock-rate on the caps
- * 734443 : qtdemux: forward DISCONT from upstream to the output streams
- * 734542 : speexenc: Improve annotation of internal function
- * 734987 : udp: fix udpsrc documentation
- * 735085 : y4mencode : port y4m encoder to use GstVideoEncoder base class
- * 735378 : gstrtpjitterbuffer: requests retransmission periodically when no needed
- * 735564 : gdkpixbufdec: Error when using gdkpixbufdec with ImageFreeze element
- * 735581 : imagefreeze: Remove impossible error condition
- * 735626 : multipartdemux: caps are NULL in pad-added callback (regression)
- * 735627 : wavenc/wavparse: should support RF64 files
- * 735795 : imagefreeze: Don't call gst_caps_unref() on NULL caps
- * 735880 : imagefreeze: replace with gst_buffer_copy
- * 735950 : gdkpixbufdec: free query after use
- * 735971 : qtdemux: avdec_mjpeg does not get autoplugged for mjpeg in mov container
- * 736072 : v4l2: set min_latency for output device according to required minimum number of buffers
- * 736122 : ximagesrc: setting the screen-num property has no effect
- * 736133 : v4l2: query crop configuration after each call of S_CROP
- * 736252 : gdkpixbufdec: packetized mode logic
- * 736462 : multifile: don't bitwise OR the same flag twice
- * 736528 : udp: getting compilation error for implicit declaration of memcmp, memset
- * 736543 : matroska:OR and Bitwise OR of the same flag twice
- * 736872 : libpng: Removed redundant assignment
- * 736873 : alpha: Removed unreachable break statements
- * 736874 : audiofx: Removed unwanted variable
- * 736875 : audiofx: Removed unwanted buffer_length variable
- * 736876 : audiofx: Removed unreachable breaks, unwanted variable
- * 736878 : audioparsers: Added index check before using the index
- * 736879 : avi: Removed redundant assignment
- * 736880 : avi: Removed unwanted hdl variable
- * 736881 : deinterlace: Removed unwanted res variable
- * 736883 : dtmf: Removed unwanted structure member and assignment
- * 736884 : flv: Removed unreachable break statements
- * 736887 : goom: Clarified precedence between % and ?
- * 736888 : isomp4: Removed unreachable breaks
- * 736890 : matroska: Removed unwanted instruction
- * 736892 : rtpmanager: Removed unwanted variable and assignment
- * 736893 : rtpmanager: Removed unwanted assignment
- * 736894 : rtpmanager: Removed unwanted assignment in rtpsession
- * 736897 : videobox: duplicate assignment
- * 736903 : rtsp: Precedence in expression is not clear
- * 736986 : qtdemux: handle AAC audio without ESDS atom
- * 737095 : qtmux: subtitle muxing doesn't work
- * 737127 : interleave: interleaving does not respect the channel positions default order
- * 737359 : matroskademux: returns FLOW_FLUSHING when trying to reuse it
- * 737708 : pngdec: change parse logic
- * 737868 : rtspsrc: set stream caps on internal src TCP pads
- * 738013 : v4l2allocator: issue with import_userptr() in single-planar API when n_planes > 1
- * 738707 : gst-plugins-good fails to build on Mac OS X 10.10 Yosemite due to deprecated NSOpenGLPFAFullScreen
- * 738838 : videobox: critical error when element properties set as max/min
- * 739344 : rtpjitterbuffer: ensure rtx_retry_period > = 0
- * 739366 : imagefreeze: Handle seqnums
- * 739549 : v4l2bufferpool: fix typos in flags
- * 739566 : gdkpixbufoverlay: Fix relative-x/y and widen their range to support scolling images in/out of frame with GstController
- * 739930 : Port server-alsasrc-PCMA.py to version 1.x
- * 739975 : Seeking through some AAC file freezes my application
- * 740403 : v4l2object: reuse caps framerate if not overwritten by v4l2 device
- * 740505 : rtspsrc: segmentation fault when requesting srtp key
- * 740683 : rtspsrc: add retransmission handling for rtp
- * 740987 : Fixes to osxaudiosrc and osxaudiosink
- * 741115 : videomixer segfault when output height is smaller than input height and ypos is negative
- * 741134 : v4l2: CREATE_BUF support is broken
- * 741279 : qtmux: generating corrupted file when over 4GB
- * 741398 : rtpptdemux: errors out on invalid rtp packet, e.g. if the version check failed (0 != 2)
- * 741993 : souphttpsrc: leaking a buffer during flushing
- * 742098 : rtp: Fails rtpaux and rtpcollision tests
- * 742325 : ac3parse: requests minimum frame size that is too small
- * 742363 : v4l2object: recognize and distinguish all bayer arrangements
- * 742572 : qtdemux: EOS emitted after 10 seconds on a audio/mp4a file [REGRESSION]
- * 742661 : qtdemux: EOS in push mode when seeking in m4a
- * 743013 : v4l2bufferpool: set v4l2_buffer.field when queuing buffer in an output device
- * 743186 : v4l2object: set colorspace in caps for capture devices
- * 743407 : qtdemux: doesn't ignore data after last sample in mdat.
- * 743518 : qtdemux: dead code while calculating segment base ?
- * 743578 : qtdemux: Parse 'sidx' atom (for duration and indexing in fragmented files)
- * 743906 : quarktv: doesn't work with planes=0, fix property range accordingly
- * 744211 : interleave: assertion 'self- > func != NULL' failed
- * 744461 : pulsesink: Enhance code readability in pulsesink_query
- * 745192 : matroskademux: V_MS-VFW-FOURCC streams have DTS instead of PTS
- * 745226 : Vorbis RTP payloader metadata is slightly wrong
- * 745276 : avidemux: remove not needed code
- * 745339 : qtdemux: key_unit seek doesn't work
- * 745441 : v4l2: Detect lossed frame and warn
- * 745515 : level: infinite loop when interval is set to low values
- * 745587 : rtp: Add PLI and FIR counters to RTPSource statistics
- * 745599 : rtsp: tcp transport fails
- * 745973 : matroskademux: gst_tag_list_insert: assertion 'GST_IS_TAG_LIST (into)' failed
- * 746065 : level: outputs random values if channels==1
- * 746242 : matroskaparse: send global tags
- * 746274 : flvdemux: Less spam from no_more_pads warning
- * 746390 : qtdemux: crash while playing MPEG DASH stream
- * 746479 : rtsp: Only two second of playback with rtpsrc and test-mp4 (rtsp-server)
- * 746543 : rtpsession: Properly implement T_rr_interval and allow sending multiple early feedback packets in a row
- * 746810 : matroska: fix GValue leak when parsing tags
- * 746822 : qtdemux: segment query reports wrong values after key-unit seek
- * 746834 : v4l2sink: driver is not queried for minimum number of buffers when propose_allocation is not called
- * 747204 : audiofirfilter creates strange noise for smaller filter kernels and even default kernel
- * 747208 : rtpvp8depay: should have width/height in its caps so it can be fed to muxers
- * 747358 : rtp: RTPJitterBufferMode enum missing from gtk-doc
- * 747394 : rtpsession: Track RTX ssrc caps
- * 747554 : suppressions: silence possible valgrind false positive
- * 747595 : tests: Add test suite for alpha element
- * 747597 : smpte: Remove unused fields
- * 747863 : rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
- * 747922 : rtpjitterbuffer/rtxreceive: Don't reset the jitterbuffer if too old RTX packets arrive
- * 748022 : audiofx: fix typos in example pipelines
- * 748024 : icydemux: Fix segfault for 0-value metainterval
- * 748041 : rtpjitterbuffer: Too early requested retransmission for future packets
- * 748353 : rtspsrc: Leak of RTCP caps
- * 748436 : rtpjitterbuffer: " stats " property docs
- * 748584 : matroskademux: fix seek event leak in push mode
- * 748617 : qtdemux: fix buffer leak on EOS with stop position in push mode
- * 748627 : rtspsrc: Don't send NACKs and early RTCP in non-feedback profiles
- * 748909 : jpegdec: fix frame leaks
- * 749054 : qtdemux: Fix gst-launch pipeline in the documentation
- * 749072 : flacparse: fix buffer leak
- * 749122 : vp8enc: vp9enc: target bitrate is not working as expected
- * 749129 : rtpg726depay: add block_align to output caps
- * 749163 : po: update POTFILES.in
- * 749543 : rtpg726depay: fix input buffer memleak
- * 749544 : rtpg726pay: fix caps leak
- * 749581 : rtpbasepayload: Try harder to reuse previously configured caps values and give more preference to anything set as properties
- * 749669 : rtp: fix collection of statistic
- * 749690 : splitfilesrc: Implement binary search in find_part_for_offset
- * 749909 : matroska: overwritten value assignment
- * 750327 : rtpssrcdemux: Add support for reduce size rtcp
- * 750332 : rtpsession: Add support for reduced size rtcp
- * 743925 : osxaudiosink won't reconfigure sink caps
- * 744922 : osxaudiosrc: iOS resampling is stuttering
- * 728353 : goom2k1: code does nothing, slowly
- * 748068 : equalizer: not changing settings dynamically
- * 731352 : flv: Container timestamp is DTS not PTS
- * 732910 : v4l2src: Dectect and workaround decreasing HW timestamp
- * 737810 : payloaders: VP8 and Opus payloader should probably suppport Google Chrome encoding-names
- * 740787 : videocrop: No longer apply the new crop if caps have not changed
- * 736396 : isomp4: duplicate if else branches in atoms.c
- * 610364 : udpsrc: allocates buffers with size a lot bigger than needed
- * 739305 : souphttpsrc: log connection events at info level
- * 744213 : spectrum: assertion 'len > 0' failed
+The GStreamer team is pleased to announce another bug-fix release in the
+stable 1.x API series of your favourite cross-platform multimedia framework!
+
+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 gst-plugins-good in the download
-directory: http://gstreamer.freedesktop.org/src/gst-plugins-good/
+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
-http://cgit.freedesktop.org/gstreamer/gst-plugins-good/
+https://gitlab.freedesktop.org/gstreamer/
==== Homepage ====
-The project's website is http://gstreamer.freedesktop.org/
+The project's website is https://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
+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 bugzilla as well.
+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).
==== Developers ====
-GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
-from there (see link above).
+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.
-
-
-Contributors to this release
-
- * Aleix Conchillo Flaqué
- * Alex O'Konski
- * Ananda
- * Andrei Sarakeev
- * Antonio Ospite
- * Anuj Jaiswal
- * Arun Raghavan
- * Aurélien Zanelli
- * Benjamin Gaignard
- * Brad Smith
- * Branislav Katreniak
- * David Sansome
- * David Schleef
- * Edward Hervey
- * George Kiagiadakis
- * Guillaume Desmottes
- * Gwenole Beauchesne
- * Göran Jönsson
- * Hans de Goede
- * Henning Heinold
- * Hyunjun Ko
- * Ilya Konstantinov
- * Jan Alexander Steffens (heftig)
- * Jan Schmidt
- * Jason Litzinger
- * Jesper Larsen
- * Jimmy Ohn
- * Jonas Holmberg
- * Jose Antonio Santos Cadenas
- * Josep Torra
- * Julien Isorce
- * Jurgen Slowack
- * Krzysztof Kotlenga
- * Linus Svensson
- * Luis de Bethencourt
- * Mark Nauwelaerts
- * Matej Knopp
- * Mathieu Duponchelle
- * Matthew Waters
- * Michael Smith
- * Miguel París Díaz
- * Nicola Murino
- * Nicolas Dufresne
- * Nicolas Huet
- * Nirbheek Chauhan
- * Ognyan Tonchev
- * Olivier Crête
- * Patrick Radizi
- * Paul Hyunil
- * Peter G. Baum
- * Peter Korsgaard
- * Peter Seiderer
- * Philippe De Muyter
- * Philippe Normand
- * Piotr Drąg
- * Ramiro Polla
- * Ravi Kiran K N
- * Reynaldo H. Verdejo Pinochet
- * Sanjay NM
- * Santiago Carot-Nemesio
- * Sebastian Dröge
- * Sebastian Rasmussen
- * Simon Farnsworth
- * Sjoerd Simons
- * Srimanta Panda
- * Stefan Sauer
- * Thiago Santos
- * Thibault Saunier
- * Tim-Philipp Müller
- * Tobias Modschiedler
- * Tom Greenwood
- * Vincent Penquerc'h
- * Vineeth T M
- * Vineeth TM
- * Víctor Manuel Jáquez Leal
- * Wim Taymans
- * Youness Alaoui
- * hark
-
\ No newline at end of file
+subscribe to the gstreamer-devel list:
+
+ https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
projects / platform / upstream / gst-plugins-good.git / blobdiff