Commit graph

5 commits

Author SHA1 Message Date
prlw1
35c01eaf9c Update gst-rtsp-server to 1.10.3
Many improvements in 2 years of development. Release notes at:
https://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/NEWS?h=1.10
2017-02-11 23:28:04 +00:00
agc
203292f73e Add SHA512 digests for distfiles for net category
Problems found with existing digests:
	Package haproxy distfile haproxy-1.5.14.tar.gz
	159f5beb8fdc6b8059ae51b53dc935d91c0fb51f [recorded]
	da39a3ee5e6b4b0d3255bfef95601890afd80709 [calculated]

Problems found locating distfiles:
	Package bsddip: missing distfile bsddip-1.02.tar.Z
	Package citrix_ica: missing distfile citrix_ica-10.6.115659/en.linuxx86.tar.gz
	Package djbdns: missing distfile djbdns-1.05-test25.diff.bz2
	Package djbdns: missing distfile djbdns-cachestats.patch
	Package djbdns: missing distfile 0002-dnscache-cache-soa-records.patch
	Package gated: missing distfile gated-3-5-11.tar.gz
	Package owncloudclient: missing distfile owncloudclient-2.0.2.tar.xz
	Package poink: missing distfile poink-1.6.tar.gz
	Package ra-rtsp-proxy: missing distfile rtspd-src-1.0.0.0.tar.gz
	Package ucspi-ssl: missing distfile ucspi-ssl-0.70-ucspitls-0.1.patch
	Package waste: missing distfile waste-source.tar.gz

Otherwise, existing SHA1 digests verified and found to be the same on
the machine holding the existing distfiles (morden).  All existing
SHA1 digests retained for now as an audit trail.
2015-11-04 00:34:51 +00:00
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
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
d6b3ad35e7 Import gst-rtsp-server-1.4.1 as net/gst-rtsp-server.
gst-rtsp-server is a library on top of GStreamer for building an
RTSP server.
2014-08-31 22:25:30 +00:00