Skip to content
Snippets Groups Projects
Commit e86bbbb6 authored by Sebastian Dröge's avatar Sebastian Dröge :tea:
Browse files

Release 1.5.1

parent 08e0c79c
No related branches found
No related tags found
Loading
This diff is collapsed.
This is GStreamer RTSP Server 1.4.0
This is GStreamer RTSP Server 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 RTSP Server Library 1.4.0
Release notes for GStreamer RTSP Server Library 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.
Features of this release
Bugs fixed in this release
* 733244 : Correct misspelled words
* 732238 : Listen on the multicast group for RTP/RTCP packets
* 734546 : tests: Unref element after usage
* 736041 : Protect rtsp transport data.
* 736647 : Tunneled RTSP sessions do not always timeout as expected
* 737110 : rtsp-client: race condition when closing client connection
* 737631 : gst-rtsp-server deadlock while sending response over TCP
* 737675 : media: media_unprepare() is kind of broken
* 737690 : rtsp-client: deadlock when setting session medias to NULL
* 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
* 737829 : rtsp-server: deactivate media when shutting down from paused
* 738905 : rtsp-client: add stream transport to the context
* 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
* 740752 : add retransmission support
* 740845 : crash when reciving a rtcp after teardown but before client finalize.
* 741678 : configure: add --disable-examples switch
* 742115 : Examples: Accept a 'port' argument for running multiple instances
* 742869 : Remove URI-escaping of RTSP session-id
* 742954 : Crash when two treads are in handle_new_sample at the same time.
* 743175 : Add support for RECORD
* 743346 : When system time is increased the ongoing RTSP sessions will time out.
* 743734 : RTCP packets not sent
* 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
* 745704 : Losing the first packet
* 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
* 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
* 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
* 749845 : Client have problem to find the teardown response.
==== Download ====
......@@ -59,7 +82,34 @@ Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Applications
Contributors to this release
* Aleix Conchillo Flaqué
* Alistair Buxton
* Andreas Frisch
* Anila Balavan
* Arun Raghavan
* Branko Subasic
* Edward Hervey
* Gregor Boirie
* Göran Jönsson
* Hyunjun Ko
* Jan Schmidt
* Kent-Inge Ingesson
* Linus Svensson
* Luis de Bethencourt
* Matthew Waters
* Nicolas Dufresne
* Nirbheek Chauhan
* Ognyan Tonchev
* Olivier Crête
* Sebastian Dröge
* Sebastian Rasmussen
* Srimanta Panda
* Stefan Sauer
* Tim-Philipp Müller
* Vincent Penquerc'h
* Wim Taymans
\ No newline at end of file
......@@ -2,7 +2,7 @@ AC_PREREQ(2.69)
dnl initialize autoconf
dnl when going to/from release please set the nano (fourth number) right !
dnl releases only do Wall, cvs and prerelease does Werror too
AC_INIT([GStreamer RTSP Server Library], [1.5.0.1],
AC_INIT([GStreamer RTSP Server Library], [1.5.1],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server])
AG_GST_INIT
......@@ -56,10 +56,10 @@ dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 501, 0, 501)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.5.0.1
GSTPB_REQ=1.5.0.1
GSTPG_REQ=1.5.0.1
GSTPD_REQ=1.5.0.1
GST_REQ=1.5.1
GSTPB_REQ=1.5.1
GSTPG_REQ=1.5.1
GSTPD_REQ=1.5.1
dnl *** autotools stuff ****
......
......@@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</repository>
<release>
<Version>
<revision>1.5.1</revision>
<branch>1.5</branch>
<name></name>
<created>2015-06-07</created>
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.1.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.4.0</revision>
......
0% Loading or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment