X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=RELEASE;h=e6d67760f87bec9d1db817784b21a984bacd7a1d;hb=59691dadd8647797abad369cdfe987b3cb864dac;hp=7de2eaba59602c58f9d39faa0dabf21dee1da2de;hpb=f9207722ca8fd8dcc1e7215d8af85efe4debfdf4;p=platform%2Fupstream%2Fgst-plugins-good.git diff --git a/RELEASE b/RELEASE index 7de2eab..e6d6776 100644 --- a/RELEASE +++ b/RELEASE @@ -1,379 +1,96 @@ +This is GStreamer gst-plugins-good 1.16.2. -Release notes for GStreamer Good Plug-ins 0.10.31 "Faster" - - - -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 - - * audioparsers: propagate downstream caps constraints upstream - * ac3parse: add support for IEC 61937 alignment and conversion/switching between alignments - * ac3parse: let bsid 9 and 10 through - * auparse: implement seeking - * avidemux: fix wrong stride when inverting uncompressed video - * cairotextoverlay: add a "silent" property to skip rendering; forward new segment events - * deinterlace: add support for deinterlacing using buffer caps/flags (as set by e.g. fieldanalysis) - * deinterlace: new fieldanalysis-related properties: "locking" and "ignore-obscure" - * directsoundsink: fix negotiation/device setup: 16-bit audio is signed, 8-bit is unsigned - * effecttv: fix reverse negotiation; repair color modes in radioactv by taking rgb,bgr into account - * equalizer: also sync the parameters for the filter bands - * flacdec: better timestamp/offset handling; try upstream first for duration queries - * flacdec: send EOS when seeking after the end of file instead of failing - * flacenc: do not drop the first data buffer on the floor - * flacparse: detect when a file lies about fixed block size; ignore invalid minimum_blocksize - * flacparse: more accurate/better duration/timestamp handling - * flvdemux: better timestamp handling (negative cts, detect large pts gaps; fix discontinuity threshold check when timestamps go backwards) - * flvmux: properly determine final duration; metadata/header writing fixes - * gdkpixbufsink: fix inverted pixel-aspect-ratio info on pixbufs - * jack: add "client-name" property to jackaudiosink and jackaudiosrc - * jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg - * jpegdec: Implement upstream negotiation - * matroskademux: seeking fixes; better handling of non-finalized files - * matroskademux: better timestamp/duration handling, fix some stuttering A/V - * matroskademux: add "max-gap-time" property to make gap handling configurable - * matroskademux: UTF-8 subtitles may have markup - * matroskamux: do not use unoffical V_MJPEG codec id - * matroskamux: fix segment handling, so we actually use running time - * matroskamux: for streaming files, push tags first - * matroskamux: handle GstForceKeyUnit event - * multifile: new splitfilesrc element to read multiple files as if they were one single file - * multifilesrc: add "loop" property - * multifilesink: handle buffer lists, useful to keep groups of buffers (GOPs) in the same file - * multifilesink: add flag to cut after a force key unit event - * multifilesink: add "max-files" property - * multifilesink: add new 'max-size' mode and "max-file-size" property for switching to the next file based on size - * multifilesink: write stream-headers when switching to the next file in max-size mode - * multipartdemux: Add property to assume a single stream and emit no-more-pads - * multipartmux: Add \r\n to tail of pushed buffers - * navseek: toggle pause/play on space bar - * osxvideo: Fix leak of NSOpenGLPixelFormat object - * pcmadepay,pcmudepay: allow variable sample rate - * pngenc: increase arbitrary resolution limits - * pulse: Drop support for PA versions before 0.9.16 (1.x is recommended) - * pulse: new pulseaudiosink element to handle format changes (not autoplugged yet) - * pulsesink: add support for compressed audio format passthrough (S/PDIF, mp3-over-bluetooth) - * pulsesink: Allow writes in bigger chunks - * pulsesink: Use the extended stream API if available - * pulsesrc: add a "source-output-index" property; implement GstStreamVolume interface - * qtdemux: better fragmented support (avoid adjustment for keyframe seek; mark all audio track samples as keyframe) - * qtdemux: parse embedded ID32 tags; improve bitrate guessing/extraction - * qtdemux: push mode fixes, fix buffered streaming - * qtmux: add direct dirac mapping - * qtmux: calculate average bitrate for streams - * qtmux: fix ctts generation for streams that don't start at 0 timestamps - * qtmux: use GST_TAG_IMAGE for coverart too - * ismlmux: Use iso-fragmented as variant type (useful in connection with encodebin) - * rtph263ppay: implement getcaps following RFC 4629, picks the right annexes - * rtph263ppay: set H263-2000 if thats what the other side wants - * rtph264depay: complete merged AU on marker bit (thus reducing latency) - * rtph264depay: cope with FU-A E bit not being set (caused by buggy payloaders) - * rtph264depay: exclude NALu size from payload length on truncated packets - * rtph264pay: proxy downstream caps restrictions (converting profile-level-id from RTP caps into video/x-h264 style caps) - * rtph264pay: only set the marker bit on the last NALU of a multi-NALU access unit - * rtpjpegpay: add support for H.264 payload in MJPEG container - * rtpjpegpay: fix for "odd" resolutions not a multiple of DCTSIZE - * rtpmp4adepay: fix output buffer timestamps in case of multiple frames - * rtpmp4gdepay: improve bogus interleaved index compensating - * rtpmp4vpay: deprecated send-config property and replace by config-interval - * rtppcmapay/depay: static clock rates on static payloads, dynamic on dynamic - * rtpvrawpay,-depay: RGB video payloading/depayloading fixes - * rtpg722pay: Compensate for clockrate vs. samplerate difference - * rtpbin: allow configurable rtcp stream syncing interval - * rtpbin: new "rtcp-sync" property, alternative inter-stream syncing methods - * rtpjitterbuffer/rtpbin: relax dropping rtcp packets; misc other fixes - * rtpmanager: don't reveal the user's username, hostname or real name by default - * rtpsession: process received Full Intra Requests (FIR) - * rtpsession: add special mode to use FIR as repair as Google does - * rtpsession: send FIR requests in response to key unit requests with all-headers=TRUE - * rtpsession: always send application requested feedback in immediate mode - * rtpsession: put the PLI requests in each RTPSource - * rtpsession: wait longer to timeout SSRC collision - * rtspsrc: implement async network I/O - * rtspsrc: allow sending short RTSP requests to a server - * rtspsrc: configure rtcp interval if provided - * rtspsrc: open on play and pause when not done yet - * shout2send: send video/webm through libshout - * soup: new souphttpclientsink element - * udpsrc: drop dataless UDP packets - * v4l2: take care not to change the current format where appropriate - * v4l2src, v4l2sink: add "norm" property; default to a pixel-aspect-ratio of 1/1 - * v4l2src: do not ignore the highest frame interval or the largest resolution - * v4l2src: handle some feature query failures more gracefully - * videobox: avoid wrapping opaque to transparent - * wavenc: Allow setcaps to be called after a format was negotiated if it's compatible - * ximagesrc: add "xid" and "xname" properties to allow capturing a particular window - * ximagesrc: fallback to non-XShm mode if allocating the XShm image failed - * ximagesrc: clear flags on buffer reuse, so that flags like DISCONT aren't set accidentally - -Bugs fixed in this release - - * 668320 : rtpmanager: RTCP receiver reports reveal full user name - * 652727 : multifilesrc: add ability to loop - * 657422 : [souphttpsrc] The souphttpsrc plugin doesn't work behind a proxy that requires authentication - * 432612 : [matroskamux] doesn't handle segments correctly - * 541215 : [avimux] Dirac muxing is broken and results in A/V sync issues - * 546932 : [ximagesrc] allow recording of specific window only - * 571400 : RTSP blocks in gst_element_set_state( GST_STATE_PAUSED ) and incorrect url - * 576524 : rtpbin, jitterbuffer: add mode to support for recording RTP streams - * 586450 : [cairotextoverlay] Forward upstream events to both sinkpads - * 595055 : [pulsesrc] Should implement GstStreamVolume interface - * 605834 : directsoundsink: 16-bit audio is always signed while 8-bit is always unsigned - * 610916 : rtspsrc dosen't work under windows - * 614803 : v4l2: add tv-norm property - * 616686 : multipartdemux: add " single-stream " property to emit no-more-pads earlier - * 616936 : [matroskademux] Incorrect display of subtitles with markup - * 619548 : qtdemux: Guess bitrate if only one stream's bitrate is unknown - * 619590 : [matroskademux] Doesn't protect segment and other fields from concurrent changes from different threads - * 620186 : qtdemux: Export max bitrate for AMR-NB/-WB streams - * 622412 : [rtpmp4vpay] remove send-config parameter; obsoleted by config-interval - * 624887 : pitivi playback hangs / errors while playing mov clips on clip change - * 630456 : [ximagesrc] Fallback to non-XShm mode if image allocation fails - * 631430 : [flvdemux] Cannot play .flv to the end - * 632504 : [rtspsrc] reduce or avoid (network) hang during shutdown - * 634093 : RTSP client asks for unicast from multicast only server - * 638300 : v4l2src: make this work more than once in a row - * 639217 : udpsrc: allow skip-first-bytes of full buffer size - * 640323 : [cairotextoverlay] forward new segment events from the sink to the source - * 643847 : deinterlace: Add support for deinterlacing using buffer caps/flags - * 644151 : [multifilesink] Add option to create a new file after each GstForceKeyUnit event - * 644154 : [matroskamux] Force a new cluster after each GstForceKeyUnit event - * 644512 : [auparse] Add seeking - * 647540 : autoaudiosink picks element to use by rank, but pulsesink/alsasink/jackaudiosink are all PRIMARY - * 648312 : [v4l2sink] Unconditionally accepts video/mpegts - * 648642 : rtpsession: Ensure ssrc collisions aren't timed out immediately - * 648937 : matroskademux: avoid building index when streamable - * 649067 : v4l2src: got unexpected frame size of 262254 instead of 614400 - * 649617 : [rtp] Deadlock and other fixes for rtpssrcdemux - * 649780 : flac: seek beyond end fails instead of EOSing immediately - * 649955 : flvmux: add support for mpegversion 2, which is also AAC - * 650258 : matroskademux/matroskaparse: gst_element_query_duration returns wrong value for Matroska files - * 650313 : ac3parse: Add support for iec61937 alignment - * 650503 : [dvdemux] Broken DURATION query handling - * 650555 : [aacparse] AAC profiles needed in caps - * 650691 : [flacparse] regression playing some flac files - * 650714 : [amrparse] skips first few frames (problem in checking sync) - * 650785 : [flacparse] duration query in DEFAULT format failing with flacparse in pipeline (regression) - * 650877 : matroska: refactor code common to matroskademux and matroskaparse - * 650912 : Rare leak in qtdemux - * 650916 : REGRESSION: ssrcdemux causing FLOW_NOT_LINKED - * 650937 : deinterlace: fix parameter type in trace - * 651059 : rtspsrc: uniform unknown message handling - * 651443 : multifilesink: add next-file=max-size mode and max-file-size property - * 652195 : matroskademux: seeking in non-finalized matroska files does not work correctly - * 652286 : matroskaparse: Gstreamer-CRITICAL when changing state from PAUSED to READY - * 652467 : matroska: missing < stdio.h > include for sscanf - * 653080 : matroskamux: make check for block_duration less sensitive - * 653091 : [dv1394src] Make the internal clock thread-safe - * 653327 : configure script for gst-plugins-good selects shout2 when it's not present - * 653559 : aacparse: too greedy minimum frame size - * 653709 : [ximagesrc] sets DISCONT on half the buffers - * 654175 : matroskademux: handle blocks with duration=0 - * 654379 : matroskamux: make default framerate optional per stream - * 654583 : Immediate RTCP in rtpsession - * 654585 : rtpmp4gdepay choppy sound - * 654744 : matroskademux: fix aspect ratio if header has only onle display variable set - * 654749 : goom: unbreak build on PPC on openbsd - * 654816 : [rtspsrc] rtspsrc doesn't get eos if it's wrapped into a bin - * 655530 : Logitech B990 HD Webcam yields poor video in MJPEG mode. - * 655570 : qtdemux: assertion error when playing Apple Trailers - * 655805 : Make the extended RTSP headers optional - * 655866 : jackaudiosink: Don't call g_alloca in jack_process_cb - * 655918 : qtdemux : qtdemux_add_fragmented_samples return error. - * 656104 : v4l2src fails to correctly configure the framerate - * 656606 : crash in gst_spectrum_reset_message_data() - * 656649 : flacparse: fix off by one in frame size check - * 656734 : [aacparse] Assumes 1024 samples per frame - * 657080 : aacparse: failing test due to two buffers being dropped for one sync loss - * 657179 : pulse: New pulseaudiosink element to handle format changes - * 657376 : rtspsrc regression - * 657830 : multiudpsink: make add/remove/clear/get-stats action signals - * 658178 : udpsrc: rough error reporting when using an invalid URI - * 658305 : [souphttpsrc] can’t seek during double speed playback - * 658419 : Add FIR support to rtpsession - * 658543 : [v4l2src] Use GST_RESOURCE_ERROR_BUSY if webcam is already used - * 658546 : ac3parse: RealAudio file with AC-3 audio no longer plays - * 659009 : [matroskademux] property for configuring gap handling - * 659065 : navseek: toggle pause/play on space bar - * 659153 : matroskademux: fix stuttering A/V - * 659237 : [gstrtpbin] clock is not unreffed after finish using it - * 659242 : [matroskademux] Unexpected EOS when seeking on paused matroska file - * 659798 : Segfault when you convert with audioconvert from audio file mkv to audio file avi - * 659808 : matroskademux: misc fixes - * 659837 : matroskamux: unable to mux audio/x-raw-int,rate=8000,channels=1,endianness=1234,width=16,depth=16,signed=true - * 659943 : [ac3parse] it does not correcly check for ac3/e-ac3 switch - * 660249 : won't play wav file: invalid WAV header (no fmt at start): ID32 - * 660275 : jpegdec doesn't implement upstream negotiation - * 660294 : goom2k1: Fix mingw compiler warnings - * 660448 : videomixer2: memory leak - * 660468 : speexenc: fix calculation of filler data size - * 660481 : v4l, ximagesrc: printf format warnings - * 660969 : qtmux memleak - * 661049 : matroskademux: support seek with start_type NONE - * 661215 : flacparse: fix last frame timestamp in fixed block size mode - * 661400 : rtpg722pay: G722 rtptime too fast - * 661477 : flvdemux: negative cts causes uint overflow, resulting in sinks waiting forever - * 661841 : [edgetv] video artifacts if videorate placed after edgetv - * 661874 : aacparse fails to forward caps to encoder - * 662856 : cairotextoverlay: add a 'silent' property to skip rendering - * 663186 : taginject is not gap aware - * 663334 : gst/flv/: add amfdefs.h to noinst_HEADERS - * 663580 : v4l2src negotiation failure with weird pixel-aspect-ratios - * 664548 : matroskaparse: memleak - * 664792 : Staircase effect in M-JPEG over RTP with unaligned image dimensions.. - * 664892 : [matroskademux] Doesn't set caps properly - * 665387 : v4l2src: fix stepwise enumeration ignoring the highest values - * 665412 : matroskamux: jpeg muxing regression - * 665502 : [flvdemux] broken a/v sync for some files - * 665666 : multifilesink: GstMultiFileSinkNext not documented - * 665872 : jackaudiosink, jackaudiosrc: add " client-name " property - * 665882 : gdkpixbufsink: " pixel-aspect-ratio " is the inverse of what it should be - * 665911 : Ability to specify ignore-length in wavparse - * 666361 : playbin2: regression: visualisations don't work with pulseaudiosink - * 666583 : matroskademux: too many bus messages in streamable mode - * 666602 : ac3parse: no valid frames found before end of stream (unexpected bsid=10) - * 666644 : udpsrc: infinite loop on dataless UDP packets - * 666688 : jpedec: peer_caps leak - * 666711 : rtspsrc: hostname lookup is not thread safe - * 667419 : matroskamux memleaks - * 667818 : osxvideo: Fix leak of NSOpenGLPixelFormat object - * 667820 : rtpptdemux: Plug potential pad leak. - * 667846 : rtph264depay: Exclude NALu size from payload length on truncated packets. - * 668648 : gst-plugins-good does not compile: cairo cannot find libgstvideo-0.10 - * 669455 : V4l2src can't open webcamstudio new vloopback - * 669590 : [shout2send] support webm streaming - * 670197 : v4l2src: webcam doesn't work due to fatal error when querying color balance attributes - * 650960 : flacparse makes decoded flac files start at sample offset 9215 - * 659947 : souphttpsink: rename to souphttpclientsink? - * 658659 : qtmux: Fix ctts entries for streams that don't start with timestamps from 0 - -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 - - * Alessandro Decina - * Alexey Fisher - * Andoni Morales Alastruey - * Antoine Jacoutot - * Arun Raghavan - * Branko Subasic - * Brian Li - * Chad - * David Henningsson - * David Schleef - * David Svensson Fors - * Debarshi Ray - * Edward Hervey - * Gary Ching-Pang Lin - * Guillaume Desmottes - * Ha Nguyen - * Havard Graff - * Jan Schmidt - * Jayakrishnan M - * John Ogness - * Jonas Larsson - * Jonny Lamb - * Julien Isorce - * Konstantin Miller - * Lasse Laukkanen - * Marc Leeman - * Mark Nauwelaerts - * Mart Raudsepp - * Miguel Angel Cabrera Moya - * Monty Montgomery - * Nicola Murino - * Nicolas Baron - * Olivier Crête - * Pascal Buhler - * Peter Korsgaard - * Peter Seiderer - * Philip Jägenstedt - * Philippe Normand - * Raimo Järvi - * Ralph Giles - * Raul Gutierrez Segales - * René Stadler - * Reynaldo H. Verdejo Pinochet - * Robert Krakora - * Sebastian Dröge - * Sebastian Rasmussen - * Sjoerd Simons - * Stas Sergeev - * Stefan Kost - * Stefan Sauer - * Stig Sandnes - * Thiago Santos - * Tim-Philipp Müller - * Tristan Matthews - * Tuukka Pasanen - * Vincent Penquerc'h - * Wim Taymans -  \ No newline at end of file +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 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://gitlab.freedesktop.org/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