Commit graph

12 commits

Author SHA1 Message Date
wiz
9ec1a0fbb5 Update to 1.4.5. All pkgsrc patches were integrated :-)
GStreamer core:
      * 736969 : queue2: dead lock when buffering
      * 738092 : basesink: clamp reported position based on direction
      * 740001 : task: race condition when pausing and stopping

GStreamer Plugins Base:
      * 741420 : video pools: should update size in configuration after applying alignment
      * 715050 : add typefinder for audio/x-audible
      * 739544 : tcp: Add test and fix memory leak in tcp elements
      * 739840 : typefind should recognize Apple Core Audio Format (CAF)
      * 740556 : videodecoder: don't complain when DTS != PTS on keyframes
      * 740675 : playsink: continues playback, reset mute property
      * 740730 : rtspconnection: don't remove child source if parent source is already destroyed
      * 740853 : audiodecoder: Push pending events before sending EOS.
      * 740952 : alsa: NetBSD fixes
      * 741045 : audiorate can can lose timestamp precision in some cases
      * 741198 : playbin: leaks GstPads

GStreamer Plugins Good:
      * 711437 : apev2mux: should not put APEv2 tags at beginning of WavPack files
      * 726194 : v4l2src does not cope well when a capture card is sometimes interlaced, sometimes progressive at the same resolution
      * 736397 : flvdemux: Per-stream resyncing behavior causes A/V-desyncs
      * 737603 : v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED
      * 739476 : vpx: fails to build against libvpx from git
      * 739722 : matroskamux: Thread safe register GstMatroskamuxPad
      * 739789 : v4l2allocator: fix error message if allocator is already active
      * 739791 : v4l2bufferpool: unref pool when v4l2_allocator_new() fails
      * 739792 : v4l2allocator: fix gst_v4l2_allocator_stop prototype
      * 739996 : videomixer: Drops a lot of frames, if one of the sources is live
      * 740040 : v4l2src: Fails in presence of V4L2_BUF_FLAG_ERROR
      * 740392 : rtspsrc: mikey base64 decoded key-mgmt leak
      * 740407 : qtmux limits capture to 4096x4096
      * 740633 : v4l2src: RW io-mode is broken
      * 740636 : v4l2src: framerate is not always set on driver
      * 740671 : aspectratiocrop: crop needs to be reset when video size changes
      * 740905 : v4l2: still has 1 include to linux/videodev.h
      * 741271 : rtph264pay: Buffer leak in H.264 payloader when using SPS/PPS
      * 741381 : rtph264pay: Race condition may cause crash when going from PAUSED- > READY
      * 741407 : deinterlace: in query_caps return only supported formats if filter is interlaced
      * 737579 : v4l2object: set colorspace for output devices
      * 739754 : v4l2bufferpool: Should validate that all memories are writeable before queueing back

GStreamer Plugins Bad:
      * 722764 : rawparse: fix SEEKING query handling
      * 729768 : mpegtsbase: Raise limit to read more bytes initially to find PCR
      * 739152 : gl/cocoa: build with GNUStep fails
      * 740191 : dvbbasesink: segfaults on 32-bit (rpi)
      * 740201 : gl/cocoa: Does not compile on OSX < 10.7 anymore
      * 740451 : srtpdec: leaks rtp/rtcp sink events
      * 740953 : configure.ac: unportable test(1) comparison operator
      * 741321 : opusparse: fix header parsing esp. of encoded output of libopus

GStreamer RTSP Server:
      * 739481 : rtsp-stream: leaks srtp decoder when leaving rtpbin
2015-01-03 18:09:30 +00:00
wiz
320a91d4fe Update to 1.4.4:
GStreamer Plugins Base:
      * 736969 : queue2: dead lock when buffering
      * 737055 : audiosink: Setting URI on playbin at about-to-finish when playing AAC and using an alsasink causes delayed playback
      * 737706 : videoencoder: release frame in finish_frame when no output state is configured
      * 737742 : vorbisdec: Crashes when handling more than 8 channels
      * 737752 : rtsp-client: crash when cleaning up session
      * 738064 : decodebin: The “drained” signal is emitted multiple times, first time too early (~1s)

GStreamer Plugins Good:
      * 726329 : vp8enc: Add support for caps renegotiation
      * 736071 : audiobasesink: Don't hold object lock while calling into other objects like the clock
      * 737735 : wavenc writes broken file if caps are set
      * 737739 : souphttpclientsink: Restarting after error results in buffers being queued forever
      * 737761 : aacparse: memory leak when converting to adts
      * 737771 : souphttpclientsink: Stream header buffer lifetime assumptions are incorrect
      * 737886 : equalizer: crash when changing equalizer settings during playback
      * 738102 : v4l2bufferpool: cleanly handle streamon failure for output device
      * 738152 : v4l2sink: leak with output device
      * 738297 : DTMF telephone-event timestamps are bogus
      * 738722 : rtpmux returns EMPTY caps when query'ing
      * 738793 : speex: encoder/decoder segfault when resetting multiple times
      * 739430 : rtspsrc: mikey related memory leaks

GStreamer Plugins Bad:
      * 732239 : h264parse: expose parsed profiles to downstream
      * 733510 : gltransformation produced black screen
      * 734156 : androidmedia: doesn't calculate framesize for COLOR_FormatYUV420Planar correctly
      * 736319 : dashdemux: mark first buffer as discont after restarting a download task
      * 737186 : h264parse: Return flushing if we get chained while being set to READY
      * 737569 : tsdemux: valid data is discarded if PES start packet is the first packet after discontinuity
      * 737658 : fluiddec: segmentation fault when used with fakesrc
      * 737724 : vc1parse: unref caps when it is empty in renegotiate()
      * 738067 : gl: Downloading YUY2 is broken and creates blocky artefacts
      * 738223 : fluiddec: leaks memory in gst_fluid_dec_change_state()
      * 738230 : vc1parser: fix level value for simple/main profile
      * 738243 : vc1parse: fix framesize when input is frame-layer
      * 738291 : fluiddec: leaks incoming caps event
      * 738449 : vc1parse: just assume none header-format when no codec_data is present
      * 738519 : vc1parse: parse frame header when stream format is ASF/raw for simple/main profile
      * 738532 : vc1parse: select caps according to wmv format at negotiation
      * 738674 : rtmpsink: leaking URI string
      * 738695 : mpegtsbase: do not remove programs on EOS
      * 738696 : hlsdemux: send missing stream start
      * 739277 : GstGLFilter propose allocation pass uninitialized size to gst_query_add_allocation_pool
      * 739348 : configure.ac: auto decision to include GL library fails
      * 739368 : gl: small memory leak in gl shader
      * 739374 : h264parse: sets srccaps too often
2014-11-23 15:54:00 +00:00
wiz
7907cf5961 Update gstreamer to 1.4.3:
Note that this announcement includes everything from 1.4.2 too, which was
never officially released as some critical bugs were found.

Bug reports fixed in this release:

GStreamer core:
      * 734412 : multiqueue: The buffering logic can lead to a pipeline stuck in PAUSED forever
      * 735574 : buffer: do not touch memory tag flag when copying buffer flags
      * 736295 : multiqueue: posts buffering message holding lock
      * 736424 : query: add annotations to gst_query_set_nth_allocation_pool
      * 736680 : basesrc: possible pool and allocator leak in prepare_allocation()
      * 736736 : query: add annotations to gst_query_add_allocation_pool
      * 736813 : typefindelement leaks sticky events upon flush_stop
      * 737102 : queue: Do not hold GST_QUEUE_LOCK while posting ERROR messages
      * 737133 : Missing gstconfig.h include

GStreamer Plugins Base:
      * 732908 : audioresample: skips samples unless input buffers have correct size
      * 727255 : playbin: filter out buffering messages when switching URI and the previous URI is still playing
      * 729811 : output-selector: test example in gstreamer plugin base is giving " Internal data flow error "
      * 735569 : rtspconnection: Crash due to no protection of watchs readsrc
      * 735748 : playbin: can't play an avi file in totem with audio-filter=scaletempo
      * 735800 : textoverlay: Two textoverlay in sequence fail to negotiate (regression)
      * 735844 : basetextoverlay/pango: overlay negotiation fails when it should not
      * 735952 : videorate: GstStructure refcount critical message
      * 736071 : audiobasesink: Don't hold object lock while calling into other objects like the clock
      * 736118 : videofilter: The buffer is not writable in transform_frame_ip
      * 736739 : audiocdsrc: do not leak uid after parsing TOC select event
      * 736779 : typefind: h265 IRAP picture always true
      * 736788 : audiodecoder: leaks events
      * 736796 : videoencoder: do not leak events when flushing them
      * 736861 : playbin: Reference count bug
      * 736679 : videodecoder: do not leak pool and allocator in error case
      * 736969 : queue2: dead lock when buffering
      * 709868 : Keep still meaningfull pending events on FLUSH_STOP

GStreamer Plugins Good:
      * 719359 : vp8dec: Doesn't handle changes in resolution
      * 733607 : v4l2transform: Rank should have been NONE
      * 734266 : vp8dec: fails when input format changes
      * 735520 : aacparse: skip valid ADTS/LOAS frames
      * 735804 : smpte: Creates incomplete raw video caps
      * 735833 : matroskademux: parse error at end of file
      * 735859 : videomixer: Dynamically changing the FPS leads to an incorrect buffer time
      * 736192 : avidemux: some AVI files crash (regression)
      * 736266 : wavparse: error in reading adtl chunk
      * 736384 : v4l2sink: pool not unreffed after usage
      * 736670 : v4l2allocator: O_CLOEXEC needs _GNU_SOURCE
      * 736805 : multipartdemux leaks new stream events
      * 736807 : rtpbin: pad leaked in error case
      * 735660 : v4l2: fix new v4l2 code not working with certain devices (regression)
      * 736944 : videoscale: vs_image_scale_4tap_Y offset should use stride to calculate buffer offset
      * 737219 : flacparse:  When generating headers, leave total_samples at 0 if upstream duration query returns GST_CLOCK_TIME_NONE.

GStreamer Plugins Bad:
      * 735861 : dataurisrc: make src thread safe
      * 736090 : aiffparse: duplicate else-if condition
      * 736390 : tsdemux: plug for a memory leak
      * 736426 : mpegpsmux: memory leak with h264/avc stream
      * 736474 : vc1parse: malformed sequence layer header and STRUCT_C
      * 736490 : tsdemux: fix overflow of packet_length field of PESHeader
      * 736729 : glmixer: do not leak pool in error cases
      * 736730 : gltestsrc: do not leak pool in error cases
      * 736731 : openni2src: do not leak pool
      * 736732 : glfilter: do not leak pool in error cases
      * 736733 : vdpdecoder: do not leak pool
      * 736735 : waylandsink: do not leak buffer pool in error case
      * 736750 : vc1parse: fix sequence-layer/frame-layer endianness
      * 736871 : codecparsers_vc1: sequence-layer parser is broken due to endianness issue.
      * 736919 : hlsdemux: attempt to unlock an already unlocked mutex in gst_hls_demux_change_playlist
      * 736951 : vc1parse: initialize sent_codec_tag before using it

GStreamer Plugins Ugly:
      * 736060 : asfdemux: add GUID for ASF_Metadata_Library_Object

GStreamer libav Plugins:
      * 734661 : avviddec: After draining frames, flush the libav decoder
      * 736515 : avviddec: keep draining buffers from libav until libav says so
      * 737144 : avauddec: keep draining buffers from libav until libav says so

GStreamer RTSP Server:
      * 735570 : Race condition between close() and handle_tunnel() causing crash
      * 736017 : Sequence number is not monotonic after PAUSE command
2014-10-01 14:26:15 +00:00
wiz
11c553105a Update to 1.4.1. Changes not found. 2014-08-31 22:24:00 +00:00
wiz
d0703f83cd Update to 1.4.0:
This is GStreamer Ugly Plugins 1.4.0

Changes since 1.2:

New API:
 • GstMessageType has GST_MESSAGE_EXTENDED added. All types before
   that can be used together as a flags type as before, but from
   that message onwards the types are just counted incrementally.
   This was necessary to be able to add more message types.
   In 2.0 GstMessageType will just become an enum and not a flags
   type anymore.
 • GstDeviceMonitor for device probing, e.g. to list all available
   audio or video capture devices. This is the replacement for
   GstPropertyProbe from 0.10.
 • Events accumulate the running-time offset now when travelling
   through pads, as set by the gst_pad_set_offset() function. This
   allows to compensate for this in the QOS event for example.
 • GstBuffer has a new flag "tag-memory" that is set automatically
   when memory is added or removed to a buffer. This allows buffer
   pools to detect if they can recycle a buffer or need to reset
   it first.
 • GstToc has new API to mark GstTocEntries as loops.
 • A not-authorized resource error has been defined to notify
   applications that accessing the resource has failed because
   of missing authorization and to distinguish this case from others.
   This change is actually already in 1.2.4.
 • GstPad has a new flag "accept-intersect", that will let the default
   ACCEPT_CAPS query handler do an intersection instead of subset check.
   This is interesting for parser elements that can handle incomplete
   caps.
 • GstCollectPads has support for flushing and a default handler for
   SEEK events now.
 • New GstFlowAggregator helper object that simplifies handling of
   flow returns in elements with multiple source pads. Additionally
   GstPad now always stores the last flow return and provides an
   API to retrieve it.
 • GstSegment has new API to offset the running time by a specific
   value and this is used in GstPad to allow positive and negative
   offsets in gst_pad_set_offset() in all situations.
 • Support for h265/HEVC and VP8 has been added to the codec utils and codec
   parsers library, and was integrated into various elements.
 • API for adjusting the TLS validation of RTSP connection has been added.
 • The RTSP and SDP library has MIKEY (RFC 3830) support now, and
   there is API to distinguish between the different RTSP profiles.
 • API to access RTP time information and statistics.
 • Support for auxiliary streams was added to rtpbin.
 • Support for tiled, raw video formats has been added.
 • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
   events and merge custom tags into them consistently.
 • GstBufferPool has support for flushing now.
 • playbin/playsink has support for application provided audio and video
   filters.
 • GstDiscoverer has new and simplified API to get details about missing
   plugins and information to pass to the plugin installer.
 • The GL library was merged from gst-plugins-gl to gst-plugins-bad,
   providing a generic infrastructure for handling GL inside GStreamer
   pipelines and a plugin with some elements using these, especially
   a video sink. Supported platforms currently are Android, Cocoa (OS X),
   DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
   Wayland and EGL platforms.
   This replaces eglglessink and also is supposed to replace osxvideosink.
 • New GstAggregator base class in gst-plugins-bad. This is supposed to
   replace GstCollectPads in the future and fix long-known shortcomings
   in its API. Together with the base class some elements are provided
   already, like a videomixer (compositor).


Major changes:
 • New plugins and elements:
   ∘ v4l2videodec element for accessing hardware codecs on
     platforms that make them accessible via V4L2, e.g.
     Samsung Exynos. This comes together with major refactoring
     of the existing V4L2 elements and the corresponding
     infrastructure.
     The v4l2videodec element replaces the mfcdec element.
   ∘ New downloadbuffer element that replaces the download
     buffering feature of queue2. Compared to queue2's code
     it is much simpler and only for this single use case.
     A noteworthy new feature is that it's downloading gaps
     in the already downloaded stream parts when nothing else
     is to be downloaded.
     This is now used by playbin when download buffering is
     enabled.
   ∘ rtpstreampay and rtpstreamdepay elements for transmitting
     RTP packets over a stream API (e.g. TCP) according to
     RFC 4571.
   ∘ rtprtx elements for standard compliant implementation of
     retransmissions, integrated into the rtpmanager plugin.
   ∘ audiomixer element that mixes multiple audio streams together
     into a single one while keeping synchronization. This is
     planned to become the replacement of the adder element.
   ∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
   ∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
   ∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
   ∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
   ∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
     are available on OS X and iOS now.

 • Other changes:
   ∘ gst-libav now uses libav 10.2, and gained support for H265/HEVC.
   ∘ Support for hardware codecs and special memory types has been
     improved with bugfixes and feature additions in various plugins
     and base classes.
   ∘ Various bugfixes and improvements to buffering in queue2 and
     multiqueue elements.
   ∘ dvbsrc supports more delivery mechanisms and other features
     now, including DVB S2 and T2 support.
   ∘ The MPEGTS library has support for many more descriptors.
   ∘ Major improvements to tsdemux and tsparse, especially time and
     seeking related.
   ∘ souphttpsrc now has support for keep-alive connections,
     compression, configurable number of retries and configuration
     for SSL certificate validation.
   ∘ hlsdemux has undergone major refactoring and works more
     reliable now and supports more HLS features like trick modes.
     Also fragments are pushed downstream while they're downloaded
     now instead of waiting for each fragment to finish.
   ∘ dashdemux and mssdemux are now also pushing fragments downstream
     while they're downloaded instead of waiting for each fragment to
     finish.
   ∘ videoflip can automatically flip based on the orientation tag.
   ∘ openjpeg supports the OpenJPEG2 API.
   ∘ waylandsink was refactored and should be more useful now. It also
     includes a small library which most likely is going to be removed
     in the future and will result in extensions to the GstVideoOverlay
     interface.
   ∘ gst-rtsp-server supports SRTP and MIKEY now.
   ∘ gst-libav encoders are now negotiating any profile/level settings
     with downstream via caps.
   ∘ Lots of fixes for coverity warnings all over the place.
   ∘ Negotiation related performance improvements.
   ∘ 800+ fixed bug reports, and many other bug fixes and other
     improvements everywhere that had no bug report.

Things to look out for:
 • The eglglessink element was removed and replaced by the glimagesink
   element.
 • The mfcdec element was removed and replaced by v4l2videodec.
 • osxvideosink is only available in OS X 10.6 or newer.
 • On Android the namespace of the automatically generated Java class
   for initialization of GStreamer has changed from com.gstreamer to
   org.freedesktop.gstreamer to prevent namespace pollution.
 • On iOS you have to update your gst_ios_init.h and gst_ios_init.m in
   your projects from the one included in the binaries if you used the
   GnuTLS GIO module before. The loading mechanism has slightly changed.
2014-08-08 21:29:39 +00:00
drochner
95f34bbf4d update to 1.0.10
changes: bugfixes
2013-12-04 11:32:13 +00:00
drochner
78c6bc7af4 update to 1.0.8
change: lamemp3enc: fix timestamping of outgoing buffers
2013-07-22 14:32:47 +00:00
drochner
13398e19ee update to 1.0.7
changes: bugfixes
2013-05-03 15:58:35 +00:00
drochner
d72b4d0079 update to 1.0.6
changes: bugfixes
2013-04-08 17:11:33 +00:00
rodent
a0a1f2e57c Fixes:
COMMENT should not be longer than 70 characters.
 COMMENT should not begin with 'A'.
 COMMENT should not begin with 'An'.
 COMMENT should not begin with 'a'.
 COMMENT should not end with a period.
 COMMENT should start with a capital letter.

pkglint warnings. Some files also got minor formatting, spelling, and style
corrections.
2013-04-06 03:45:05 +00:00
drochner
f229217cb4 update to 1.0.5
changes: bugfixes
2013-03-15 18:34:46 +00:00
ryoon
16b390e308 Import gst-plugins1-ugly-1.0.3 as multimedia/gst-plugins1-ugly.
GStreamer is a library that allows the construction of graphs of
media-handling components, ranging from simple Ogg/Vorbis playback to
complex audio (mixing) and video (non-linear editing) processing.

Applications can take advantage of advances in codec and filter technology
transparently.  Developers can add new codecs and filters by writing a
simple plugin with a clean, generic interface.

GStreamer is released under the LGPL.

This package is part of the ugly GStreamer plugins; that is, those that
might pose some legal problems.
2012-11-29 08:27:25 +00:00