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| Release notes for GStreamer Good Plugins 1.5.1 |
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| 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. |
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| |
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| "Such ingratitude. After all the times I've saved your life." |
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| 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) |
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| Bugs fixed in this release |
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| * 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 |
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| ==== Download ==== |
| |
| You can find source releases of gst-plugins-good in the download |
| directory: http://gstreamer.freedesktop.org/src/gst-plugins-good/ |
| |
| The git repository and details how to clone it can be found at |
| http://cgit.freedesktop.org/gstreamer/gst-plugins-good/ |
| |
| ==== Homepage ==== |
| |
| The project's website is 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 |
| |
| Please submit patches via bugzilla as well. |
| |
| 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 is stored in Git, hosted at git.freedesktop.org, and can be cloned |
| from there (see link above). |
| |
| 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 |
| |
