Commit a7faa3e0 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.5.1

parent 5c93af74
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This is GStreamer Good Plugins 1.4.0
This is GStreamer Good Plugins 1.5.1
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.
Release notes for GStreamer Good Plugins 1.4.0
Release notes for GStreamer Good Plugins 1.5.1
The GStreamer team is pleased to announce the first release of
the stable 1.4 release series. The 1.4 release series is adding new
features on top of the 1.0 and 1.2 series and is part of the API and
ABI-stable 1.x release series of the GStreamer multimedia framework.
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 are provided together
with this release.
The stable 1.4 release series is API and ABI compatible with 1.0.x,
1.2.x and any other 1.x release series in the future. Compared to 1.2.x
it contains some new features and more intrusive changes that were
considered too risky as a bugfix.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.5 release series.
......@@ -64,10 +58,172 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 733122 : rgvolume/rtpdtmfmux: Avoid taking unnecessary ref to pad templates.
* 733190 : [regression] aacparse: raw to ADTS conversion no longer works
* 733208 : POTFILES.in is out of date
* 733380 : videobox: adds borders of the wrong color
* 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
==== Download ====
......@@ -104,8 +260,82 @@ 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
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
AC_INIT([GStreamer Good Plug-ins],[1.5.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AC_INIT([GStreamer Good Plug-ins],[1.5.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
......@@ -46,8 +46,8 @@ AG_GST_LIBTOOL_PREPARE
AS_LIBTOOL(GST, 501, 0, 501)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.5.0.1
GSTPB_REQ=1.5.0.1
GST_REQ=1.5.1
GSTPB_REQ=1.5.1
dnl *** autotools stuff ****
......
......@@ -198,6 +198,26 @@
<DEFAULT>2147483648</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiFileSink::aggregate-gops</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Aggregate GOPs</NICK>
<BLURB>Whether to aggregate GOPs and process them as a whole without splitting.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiFileSink::max-file-duration</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum File Duration</NICK>
<BLURB>Maximum file duration before starting a new file in max-size mode.</BLURB>
<DEFAULT>18446744073709551615</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiFileSrc::caps</NAME>
<TYPE>GstCaps*</TYPE>
......@@ -1151,7 +1171,7 @@
<ARG>
<NAME>GstQuarkTV::planes</NAME>
<TYPE>gint</TYPE>
<RANGE>[0,64]</RANGE>
<RANGE>[1,64]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Planes</NICK>
<BLURB>Number of planes.</BLURB>
......@@ -21358,6 +21378,16 @@
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpBin::rtp-profile</NAME>
<TYPE>GstRTPProfile</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>RTP Profile</NICK>
<BLURB>Default RTP profile of newly created sessions.</BLURB>
<DEFAULT>GST_RTP_PROFILE_AVPF</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJitterBuffer::do-lost</NAME>
<TYPE>gboolean</TYPE>
......@@ -21498,6 +21528,26 @@
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJitterBuffer::rtx-max-retries</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>RTX Max Retries</NICK>
<BLURB>The maximum number of retries to request a retransmission. (-1 not limited).</BLURB>
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJitterBuffer::rtx-next-seqnum</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>RTX next seqnum</NICK>
<BLURB>Estimate when the next packet should arrive and schedule a retransmission request for it.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpSession::bandwidth</NAME>
<TYPE>gdouble</TYPE>
......@@ -21505,7 +21555,7 @@
<FLAGS>rw</FLAGS>
<NICK>Bandwidth</NICK>
<BLURB>The bandwidth of the session in bytes per second (0 for auto-discover).</BLURB>
<DEFAULT>64000</DEFAULT>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
......@@ -21555,7 +21605,7 @@
<FLAGS>rw</FLAGS>
<NICK>RTCP Fraction</NICK>
<BLURB>The RTCP bandwidth of the session in bytes per second (or as a real fraction of the RTP bandwidth if < 1.0).</BLURB>
<DEFAULT>3200</DEFAULT>
<DEFAULT>0.05</DEFAULT>
</ARG>
<ARG>
......@@ -21628,6 +21678,16 @@
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpSession::rtp-profile</NAME>
<TYPE>GstRTPProfile</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>RTP Profile</NICK>
<BLURB>RTP profile to use.</BLURB>
<DEFAULT>GST_RTP_PROFILE_AVP</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxSend::rtx-payload-type</NAME>
<TYPE>guint</TYPE>
......
......@@ -33,6 +33,10 @@ GObject
GstMuLawEnc
GstSpeexEnc
GstWavpackEnc
GstAudioVisualizer-ExtGom
GstGoom
GstAudioVisualizer-ExtGoom2k1
GstGoom2k1
GstAviDemux
GstAviMux
GstAviSubtitle
......@@ -164,8 +168,6 @@ GObject
GstFlvMux
GstFlxDec
GstGdkPixbufDec
GstGoom
GstGoom2k1
GstICYDemux
GstISMLMux
GstImageFreeze
......
......@@ -3,10 +3,10 @@
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>adds an alpha channel to video - constant or via chroma-keying</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Parsers for various audio formats</description>
<filename>../../gst/audioparsers/.libs/libgstaudioparsers.so</filename>
<basename>libgstaudioparsers.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
<filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
<basename>libgstautodetect.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
<version>1.5.0.1</version>
<version>1.5.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
...