Hello community,
here is the log from the commit of package gstreamer-rtsp-server for
openSUSE:Factory checked in at 2019-10-11 15:14:11
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-rtsp-server (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.2352 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-rtsp-server"
Fri Oct 11 15:14:11 2019 rev:22 rq:733429 version:1.16.1
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-rtsp-server/gstreamer-rtsp-server.changes
2019-06-30 14:41:01.075947685 +0200
+++
/work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.2352/gstreamer-rtsp-server.changes
2019-10-11 15:14:25.752579933 +0200
@@ -1,0 +2,6 @@
+Tue Sep 24 15:01:29 UTC 2019 - Bjørn Lie <[email protected]>
+
+- Update to version 1.16.1:
+ + See main gstreamer package for changelog.
+
+-------------------------------------------------------------------
Old:
----
gst-rtsp-server-1.16.0.tar.xz
New:
----
gst-rtsp-server-1.16.1.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-rtsp-server.spec ++++++
--- /var/tmp/diff_new_pack.oPZsiL/_old 2019-10-11 15:14:27.708574556 +0200
+++ /var/tmp/diff_new_pack.oPZsiL/_new 2019-10-11 15:14:27.712574545 +0200
@@ -18,7 +18,7 @@
%define _name gst-rtsp-server
Name: gstreamer-rtsp-server
-Version: 1.16.0
+Version: 1.16.1
Release: 0
Summary: GStreamer-based RTSP server library
License: LGPL-2.0-or-later
@@ -27,7 +27,7 @@
Source0:
https://gstreamer.freedesktop.org/src/gst-rtsp-server/%{_name}-%{version}.tar.xz
Source99: gstreamer-rtsp-server-rpmlintrc
-BuildRequires: meson
+BuildRequires: meson >= 0.47
BuildRequires: pkgconfig
BuildRequires: pkgconfig(glib-2.0) >= 2.40.0
BuildRequires: pkgconfig(gobject-introspection-1.0) >= 1.31.1
++++++ gst-rtsp-server-1.16.0.tar.xz -> gst-rtsp-server-1.16.1.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/ChangeLog
new/gst-rtsp-server-1.16.1/ChangeLog
--- old/gst-rtsp-server-1.16.0/ChangeLog 2019-04-19 01:34:55.000000000
+0200
+++ new/gst-rtsp-server-1.16.1/ChangeLog 2019-09-23 12:17:42.000000000
+0200
@@ -1,3 +1,55 @@
+=== release 1.16.1 ===
+
+2019-09-23 11:17:41 +0100 Tim-Philipp Müller <[email protected]>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.16.1
+
+2019-08-30 14:00:52 +0200 Kristofer Björkström <[email protected]>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: RTP Info must exist in PLAY response
+ If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
+ Fixes #76
+
+2019-06-25 13:19:44 +0100 Tim-Philipp Müller <[email protected]>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ onvif-media: fix "void function returning a value" compiler warning
+
+2019-04-23 14:38:05 +0300 Sebastian Dröge <[email protected]>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Add various missing Since: 1.16 markers
+
+2019-04-23 15:01:32 +0300 Sebastian Dröge <[email protected]>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Add various Since: 1.14 markers
+
+2019-04-23 15:09:34 +0300 Sebastian Dröge <[email protected]>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-server: Fix various Since markers
+
+2019-05-02 12:35:34 +0100 Tim-Philipp Müller <[email protected]>
+
+ * .gitlab-ci.yml:
+ ci: use template from 1.16 branch
+
=== release 1.16.0 ===
2019-04-19 00:34:54 +0100 Tim-Philipp Müller <[email protected]>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/NEWS
new/gst-rtsp-server-1.16.1/NEWS
--- old/gst-rtsp-server-1.16.0/NEWS 2019-04-19 01:34:54.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/NEWS 2019-09-23 12:17:41.000000000 +0200
@@ -5,10 +5,13 @@
GStreamer 1.16.0 was originally released on 19 April 2019.
+The latest bug-fix release in the 1.16 series is 1.16.1 and was released
+on 23 September 2019.
+
See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
version of this document.
-_Last updated: Friday 19 April 2019, 00:00 UTC (log)_
+_Last updated: Sunday 22 September 2019, 21:00 UTC (log)_
Introduction
@@ -353,7 +356,7 @@
- rtpjitterbuffer has improved end-of-stream handling
-- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
+- rtpmp4vpay will be preferred over rtpmp4gpay for MPEG-4 video in
autoplugging scenarios now
- rtspsrc now allows applications to send RTSP SET_PARAMETER and
@@ -1208,7 +1211,7 @@
used in order to re-produce a specific build. To set a manifest, you
can set manifest = 'my_manifest.xml' in your configuration file, or
use the --manifest command line option. The command line option will
- take precendence over anything specific in the configuration file.
+ take precedence over anything specific in the configuration file.
- The new build-deps command can be used to build only the
dependencies of a recipe, without the recipe itself.
@@ -1224,6 +1227,12 @@
section in the Cerbero documentation for more details how to enable
and use these variants.
+- When building on Windows, Cerbero can now build GStreamer recipes
+ and core dependencies such as glib with Visual Studio. This is
+ controlled by the visualstudio variant. Visual Studio 2015, 2017,
+ and 2019 are supported. Currently, only 64-bit x86 is supported due
+ to a known bug which will be fixed for the next release.
+
- A new -t / --timestamp command line switch makes commands print
timestamps
@@ -1353,6 +1362,342 @@
1.16.0 was released on 19 April 2019.
+1.16.1
+
+The first 1.16 bug-fix release (1.16.1) was released on 23 September
+2019.
+
+This release only contains bugfixes and it _should_ be safe to update
+from 1.16.0.
+
+Highlighted bugfixes in 1.16.1
+
+- GStreamer-vaapi: fix green frames and decoding artefacts in some
+ cases
+- OpenGL: fix wayland event source burning CPU in certain
+ circumstances
+- Memory leak fixes and memory footprint improvements
+- Performance improvements
+- Stability and security fixes
+- Fix enum for GST_MESSAGE_DEVICE_CHANGED which is technically an API
+ break, but this is only used internally in GStreamer and duplicated
+ another message enum
+- hls: Make crypto dependency optional when hls-crypto is auto
+- player: fix switching back and forth between forward and reverse
+ playback
+- decklinkaudiosink: Drop late buffers
+- openh264enc: Fix compilation with openh264 v2.0
+- wasapisrc: fix segtotal value being always 2
+- Fix issues on Android Q
+
+gstreamer core
+
+- device: gst_device_create_element() is transfer floating, not
+ transfer full
+- filesink, fdsink: respect IOV_MAX for the writev iovec array
+ (Solaris)
+- miniobject: free qdata array when the last qdata is removed (reduces
+ memory footprint)
+- bin: Fix minor race when adding to a bin
+- aggregator: Actually handle NEED_DATA return from update_src_caps()
+- aggregator: Ensure that the source pad is created as a
+ GstAggregatorPad if no type is given in the pad template
+- latency: fix custom event leaks
+- registry: Use plugin directory from the build system for
+ relocateable Windows builds
+- message: fix up enum value for GST_MESSAGE_DEVICE_CHANGED
+- info: Fix deadlock in gst_ring_buffer_logger_log()
+- downloadbuffer: Check for flush after seek
+- identity: Non-live upstream have no max latency
+- identity: Fix the ts-offset property getter
+- aggregator: Make parsing of explicit sink pad names more robust
+- bufferpool: Fix the buffer size reset code
+- fakesink, fakesrc, identity: sync gst_buffer_get_flags_string() with
+ new flags
+- multiqueue: never unref queries we do not own
+- concat: Reset last_stop on FLUSH_STOP too
+- aggregator: fix flow-return boolean return type mismatch
+- gstpad: Handle probes that reset the data field
+- gst: Add support for g_autoptr(GstPromise)
+- gst-inspect: fix unused-const-variable error in windows
+- base: Include gstbitwriter.h in the single-include header
+- Add various Since: 1.16 markers
+- GST_MESSAGE_DEVICE_CHANGED duplicates GST_MESSAGE_REDIRECT
+- Targetting wrong meson version
+- meson: Make get_flex_version.py script executable
+- meson: Link to objects instead of static helper library
+- meson: set correct install path for gdb helper
+- meson: fix warning about configure_file() install kwarg
+
+gst-plugins-base
+
+- video-info: parse field-order for all interleaved formats
+- tests: fix up valgrind suppressions for glibc getaddrinfo leaks
+- meson: Reenable NEON support (in audio resampler)
+- audio-resampler: Update NEON to handle remainders not multiples of 4
+- eglimage: Fix memory leak
+- audiodecoder: Set output caps with negotiated caps to avoid critical
+ info printed
+- video-frame: Take TFF flag from the video info if it was set in
+ there
+- glcolorconvert: Fix external-oes shader
+- video-anc: Fix ADF detection when trying to extract data from vanc
+- gl/wayland: fix wayland event source burning CPU
+- configure: add used attribute in order to make NEON detection
+ working with -flto.
+- audioaggregator: Return a valid rate range from caps query if
+ downstream supports a whole range
+- rtspconnection: data-offset increase not set
+- rtpsconnection: Fix number of n_vectors
+- video-color: Add compile-time assert for ColorimetryInfo enum
+- audiodecoder: Fix leak on failed audio gaps
+- glupload: Keep track of cached EGLImage texture format
+- playsink: Set ts-offset to text sink.
+- meson.build: use join_paths() on prefix
+- compositor: copy frames as-is when possible
+- compositor: Skip background when a pad obscures it completely
+- rtspconnection: Start CSeq at 1 (some servers don’t cope well with
+ seqnum 0)
+- viv-fb: fix build break for GST_GL_API
+- gl/tests: fix shader creation tests part 2
+- gl/tests: fix shader creation tests
+- wayland: set the event queue also for the xdg_wm_base object
+- video: Added GI annotation for gstvideoaffinetransformationmeta
+ apply_matrix
+- compositor: Remove unneeded left shift for ARGB/AYUV SOURCE operator
+- Colorimetry fixes
+- alsasrc: Don’t use driver timestamp if it’s zero
+- gloverlaycompositor: fix crash if buffer doesn’t have video meta
+- meson: Don’t try to find gio-unix on Windows
+- glshader: fix default external-oes shaders
+- subparse: fix pushing WebVTT cue with no newline at the end
+- meson: Missing “android” choice in gl_winsys
+- video test: Keep BE test inline with LE test
+- id3tag: Correctly validate the year from v1 tags before passing to
+ GstDateTime
+- gl/wayland: Don’t prefix wl_shell struct field
+- eglimage: Add compatibility define for DRM_FORMAT_NV24
+- Add various Since: 1.16 markers
+- video-anc: Handle SD formats correctly
+- Docs: add GL_CFLAGS to GTK_DOC_CFLAGS
+- GL: using vaapi and showing on glimagesink on wayland loads one core
+ for 100% on 1.16
+- GL: external-oes shader places precision qualifier before #extension
+ (was: androidmedia amcviddec fail after 1.15.90 1.16.0 update)
+
+gst-plugins-good
+
+- alpha: Fix one_over_kc calculation on arm/aarch64
+- souphttpsrc: Fix incompatible type build warning
+- rtpjitterbuffer: limit max-dropout-time to maxint32
+- rtpjitterbuffer: Clear clock master before unreffing
+- qtdemux: Use empty-array safe way to cleanup GPtrArray
+- v4l2: Fix type compatibility issue with glibc 2.30
+- valgrind: suppress Cond error coming from gnutls and Ignore leaks
+ caused by shout/sethostent
+- rtpfunnel: forward correct segment when switching pad
+- gtkglsink: fix crash when widget is resized after element
+ destruction
+- jpegdec: Don’t dereference NULL input state if we have no caps in
+ TIME segments
+- rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps
+- v4l2videodec: return right type for drain.
+- rtpssrcdemux: Avoid taking streamlock out-of-band
+- Support v4l2src buffer orphaning
+- splitmuxsink: Only set running time on finalizing sink element when
+ in async-finalize mode
+- rtpsession: Always keep at least one NACK on early RTCP
+- rtspsrc: do not try to send EOS with invalid seqnum
+- rtpsession: Call on-new-ssrc earlier
+- rtprawdepay: Don’t get rid of the buffer pool on FLUSH_STOP
+- rtpbin: Free storage when freeing session
+- scaletempo: Advertise interleaved layout in caps templates
+- Support v4l2src buffer orphaning
+
+gst-plugins-bad
+
+- hls: Make crypto dependency optional when hls-crypto is auto
+- player: fix switching back and forth between forward and reverse
+ playback
+- decklinkaudiosink: Drop late buffers
+- srt: Add stats property, include sender-side statistics and fix a
+ crash
+- dshowsrcwrapper: fix regression on device selection
+- tsdemux: Limit the maximum PES payload size
+- wayland: Define libdrm_dep in meson.build to fix meson configure
+ error when kms is disabled
+- sctp: Fix crash on free() when using the MSVC binaries
+- webrtc: Fix signals documentation
+- h264parse: don’t critical on VUI parameters > 2^31
+- rtmp: Fix crash inside free() with MSVC on Windows
+- iqa: fix leak of map_meta.data
+- d3dvideosink: Fix crash on WinProc handler
+- amc: Fix crash when a sync_meta survives its sink
+- pitch: Fix race between putSamples() and setting soundtouch
+ parameters
+- webrtc: fix type of max-retransmits, make it work
+- mxfdemux: Also allow picture essence element type 0x05 for VC-3
+- wasapi: fix symbol redefinition build error
+- decklinkvideosrc: Retrieve mode of the ancillary data from the frame
+- decklinkaudiosrc/decklinkvideosrc: Do nothing in
+ BaseSrc::negotiate() and…
+- adaptivedemux: do not retry downloads during shutdown.
+- webrtcbin: fix GInetAddress leak
+- dtls: fix dtls connection object leak
+- siren: fix a global buffer overflow spotted by asan
+- kmssink: Fix implicit declaration build error
+- Fix -Werror=return-type error in configure.
+- aiff: Fix infinite loop in header parsing.
+- nvdec: Fix possible frame drop on EOS
+- srtserversrc: yields malformed rtp payloads
+- srtsink: Fix crash in case no URI
+- dtlsagent: Fix leaked dtlscertificate
+- meson: bluez: Early terminate configure on Windows
+- decklink: Correctly ensure >=16 byte alignment for the buffers we
+ allocate
+- webrtcbin: fix DTLS when receivebin is set to DROP
+- zbar: Include running-time, stream-time and duration in the messages
+- uvch264src: Make sure we set our segment
+- avwait: Allow start and end timecode to be set back to NULL
+- avwait: Don’t print warnings for every buffer passed
+- hls/meson: fix dependency logic
+- Waylandsink gnome shell workaround
+- avwait: Allow setting start timecode after end timecode; protect
+ propeties with mutex
+- wayland/wlbuffer: just return if used_by_compositor is true when
+ attach
+- proxy: Set SOURCE flag on the source and SINK flag on the sink
+- ivfparse: Check the data size against IVF_FRAME_HEADER_SIZE
+- webrtc: Add various Since markers to new types after 1.14.0
+- msdk: fix the typo in debug category
+- dtlsagent: Do not overwrite openssl locking callbacks
+- meson: Fix typo in gsm header file name
+- srt: handle races in state change
+- webrtc: Add g_autoptr() support for public types
+- openh264enc: Fix compilation with openh264 v2.0
+- meson: Allow CUDA_PATH fallback on linux
+- meson: fix build with opencv=enabled and opencv4. Fixes #964
+- meson: Add support for the colormanagement plugin
+- autotools: gstsctp: set LDFLAGS
+- nvenc/nvdec: Add NVIDIA SDK headers to noinst_HEADERS
+- h264parse: Fix typo when setting multiview mode and flags
+- Add various Since: 1.16 markers
+- opencv: allow compilation against 4.1.x
+- Backport of some minor srt commits without MR into 1.16
+- meson: fix build with opencv=enabled and opencv4
+- wasapisrc: fix segtotal value being always 2 due to an unused
+ variable
+- meson: colormanagement missing
+- androidmedia amcviddec fail after 1.15.90 1.16.0 update
+
+gst-plugins-ugly
+
+- meson: Always require the gmodule dependency
+
+gst-libav
+
+- docs: don’t include the type hierarchy, fixing build with gtk-doc
+ 1.30
+- avvidenc: Correctly signal interlaced input to ffmpeg when the input
+ caps are interlaced
+- autotools: add bcrypt to win32 libs
+- gstav: Use libavcodec util function for version check
+- API documentation fails to build with gtk-doc 1.30
+
+gst-rtsp-server
+
+- rtsp-client: RTP Info must exist in PLAY response
+- onvif-media: fix “void function returning a value” compiler warning
+- Add various Since: 1.16 markers
+
+gstreamer-vaapi
+
+- fix egl context leak and display creation race
+- pluginutil: Remove Mesa from drivers white list
+- Classify vaapidecodebin as a hardware decoder
+- Fix two leak
+- vaapivideomemory: demote error message to info
+- encoder: vp8,vp9: reset frame_counter when input frame’s format
+ changes
+- encoder: mpeg2: No packed header for SPS and PPS
+- decoder: vp9: clear parser pointer after release
+- encoder: Fixes deadlock in change state function
+- encoder: h265: reset num_ref_idx_l1_active_minus1 when low delay B.
+- encoder: not call ensure_num_slices inside g_assert()
+- encoder: continue if roi meta is NULL
+- decoder: vp9: Set chroma_ ype by VP9 bit_depth
+- vaapipostproc: don’t do any color conversion when GL_TEXTURE_UPLOAD
+- libs: surface: fix double free when dmabuf export fails
+- h264 colors and artifacts upon upgrade to GStreamer Core Library
+ version 1.15.90
+
+gst-editing-services
+
+- element: Properly handle the fact that pasting can return NULL
+- Add various missing Since markers
+- launch: Fix caps restriction short names
+- python: Avoid warning about using deprecated methods
+- video-transition: When using non crossfade effect use ‘over’
+ operations
+- meson: Generate a pkgconfig file for the GES plugin
+
+gst-devtools
+
+- launcher: testsuites: skip systemclock stress tests
+- validate: fix build on macOS
+
+gst-build
+
+- Update win flex bison binaries
+- Update the flexmeson windows binary version
+- Don’t allow people to run meson inside the uninstalled env
+
+Cerbero build tool and packaging changes in 1.16.1
+
+- cerbero: Add enums for Fedora 30, Fedora 31 and Debian bullseye
+- gnutls.recipe: Fix crash when running on Android Q
+- recipes: Upgrade openssl to 1.1.1c
+- Fix some typos
+- add support for vs build tools 2019, fixes #183
+- android: Adjust gstreamer-1.0.mk for NDK r20
+- Fix license enums
+- bootstrap: Fix dnf usage on CentOS
+- Make _add_system_libs reentrant
+- meson.recipe: Fix setting of bitcode compiler options
+- cerbero: support Ubuntu disco dingo
+- cerbero: Set utf-8 to execution character set also on MSVC
+- git: simplify the reset of the source branch.
+- FORTIFY: %n not allowed on Android Q
+- Fails to build if there’s no license file for the given license
+ (GPL/LGPL without Plus, Proprietary, …)
+
+Contributors to 1.16.1
+
+Aaron Boxer, Adam Duskett, Alicia Boya García, Andoni Morales Alastruey,
+Antonio Ospite, Arun Raghavan, Askar Safin, A. Wilcox, Charlie Turner,
+Christoph Reiter, Damian Hobson-Garcia, Daniel Klamt, Danny Smith, David
+Gunzinger, David Ing, David Svensson Fors, Doug Nazar, Edward Hervey,
+Eike Hein, Fabrice Bellet, Fernando Herrrera, Georg Lippitsch, Göran
+Jönsson, Guillaume Desmottes, Haihao Xiang, Haihua Hu, Håvard Graff, Hou
+Qi, Ignacio Casal Quinteiro, Ilya Smelykh, Jan Schmidt, Javier Celaya,
+Jim Mason, Jonas Larsson, Jordan Petridis, Jose Antonio Santos Cadenas,
+Juan Navarro, Knut Andre Tidemann, Kristofer Björkström, Lucas Stach,
+Marco Felsch, Marcos Kintschner, Mark Nauwelaerts, Martin Liska, Martin
+Theriault, Mathieu Duponchelle, Matthew Waters, Michael Olbrich, Mike
+Gorse, Nicola Murino, Nicolas Dufresne, Niels De Graef, Niklas
+Hambüchen, Nirbheek Chauhan, Olivier Crête, Philippe Normand, Ross
+Burton, Sebastian Dröge, Seungha Yang, Song Bing, Thiago Santos,
+Thibault Saunier, Thomas Coldrick, Tim-Philipp Müller, Víctor Manuel
+Jáquez Leal, Vivia Nikolaidou, Xavier Claessens, Yeongjin Jeong,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.16.1
+
+- List of Merge Requests applied in 1.16
+- List of Issues fixed in 1.16.1
+
Known Issues
@@ -1376,9 +1721,9 @@
development of 1.17/1.18 will happen in the git master branch.
The plan for the 1.18 development cycle is yet to be confirmed, but it
-is possible that the next cycle will be a short one in which case
-feature freeze would be perhaps around August 2019 with a new 1.18
-stable release in September.
+is now expected that feature freeze will take place shortly after the
+GStreamer conference/hackfest in early November 2019, with the first
+1.18 stable release ready in late November or early December.
1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
1.8, 1.6, 1.4, 1.2 and 1.0 release series.
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/RELEASE
new/gst-rtsp-server-1.16.1/RELEASE
--- old/gst-rtsp-server-1.16.0/RELEASE 2019-04-19 01:34:54.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/RELEASE 2019-09-23 12:17:41.000000000 +0200
@@ -1,10 +1,7 @@
-This is GStreamer gst-rtsp-server 1.16.0.
+This is GStreamer gst-rtsp-server 1.16.1.
-The GStreamer team is thrilled to announce a new major feature release in the
-stable 1.0 API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+The GStreamer team is pleased to announce another bug-fix release in the
+stable 1.x API series of your favourite cross-platform multimedia framework!
The 1.16 release series adds new features on top of the 1.14 series and is
part of the API and ABI-stable 1.x release series of the GStreamer multimedia
@@ -60,7 +57,7 @@
directory: https://gstreamer.freedesktop.org/src/gstreamer/
The git repository and details how to clone it can be found at
-https://cgit.freedesktop.org/gstreamer/gstreamer/
+https://gitlab.freedesktop.org/gstreamer/
==== Homepage ====
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/configure
new/gst-rtsp-server-1.16.1/configure
--- old/gst-rtsp-server-1.16.0/configure 2019-04-19 01:34:23.000000000
+0200
+++ new/gst-rtsp-server-1.16.1/configure 2019-09-23 12:17:09.000000000
+0200
@@ -1,6 +1,6 @@
#! /bin/sh
# Guess values for system-dependent variables and create Makefiles.
-# Generated by GNU Autoconf 2.69 for GStreamer RTSP Server Library 1.16.0.
+# Generated by GNU Autoconf 2.69 for GStreamer RTSP Server Library 1.16.1.
#
# Report bugs to <http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer>.
#
@@ -591,8 +591,8 @@
# Identity of this package.
PACKAGE_NAME='GStreamer RTSP Server Library'
PACKAGE_TARNAME='gst-rtsp-server'
-PACKAGE_VERSION='1.16.0'
-PACKAGE_STRING='GStreamer RTSP Server Library 1.16.0'
+PACKAGE_VERSION='1.16.1'
+PACKAGE_STRING='GStreamer RTSP Server Library 1.16.1'
PACKAGE_BUGREPORT='http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer'
PACKAGE_URL=''
@@ -1537,7 +1537,7 @@
# Omit some internal or obsolete options to make the list less imposing.
# This message is too long to be a string in the A/UX 3.1 sh.
cat <<_ACEOF
-\`configure' configures GStreamer RTSP Server Library 1.16.0 to adapt to many
kinds of systems.
+\`configure' configures GStreamer RTSP Server Library 1.16.1 to adapt to many
kinds of systems.
Usage: $0 [OPTION]... [VAR=VALUE]...
@@ -1609,7 +1609,7 @@
if test -n "$ac_init_help"; then
case $ac_init_help in
- short | recursive ) echo "Configuration of GStreamer RTSP Server Library
1.16.0:";;
+ short | recursive ) echo "Configuration of GStreamer RTSP Server Library
1.16.1:";;
esac
cat <<\_ACEOF
@@ -1807,7 +1807,7 @@
test -n "$ac_init_help" && exit $ac_status
if $ac_init_version; then
cat <<\_ACEOF
-GStreamer RTSP Server Library configure 1.16.0
+GStreamer RTSP Server Library configure 1.16.1
generated by GNU Autoconf 2.69
Copyright (C) 2012 Free Software Foundation, Inc.
@@ -2085,7 +2085,7 @@
This file contains any messages produced by compilers while
running configure, to aid debugging if configure makes a mistake.
-It was created by GStreamer RTSP Server Library $as_me 1.16.0, which was
+It was created by GStreamer RTSP Server Library $as_me 1.16.1, which was
generated by GNU Autoconf 2.69. Invocation command line was
$ $0 $@
@@ -3062,7 +3062,7 @@
# Define the identity of the package.
PACKAGE='gst-rtsp-server'
- VERSION='1.16.0'
+ VERSION='1.16.1'
cat >>confdefs.h <<_ACEOF
@@ -3273,9 +3273,9 @@
- PACKAGE_VERSION_MAJOR=$(echo 1.16.0 | cut -d'.' -f1)
- PACKAGE_VERSION_MINOR=$(echo 1.16.0 | cut -d'.' -f2)
- PACKAGE_VERSION_MICRO=$(echo 1.16.0 | cut -d'.' -f3)
+ PACKAGE_VERSION_MAJOR=$(echo 1.16.1 | cut -d'.' -f1)
+ PACKAGE_VERSION_MINOR=$(echo 1.16.1 | cut -d'.' -f2)
+ PACKAGE_VERSION_MICRO=$(echo 1.16.1 | cut -d'.' -f3)
@@ -3286,7 +3286,7 @@
{ $as_echo "$as_me:${as_lineno-$LINENO}: checking nano version" >&5
$as_echo_n "checking nano version... " >&6; }
- NANO=$(echo 1.16.0 | cut -d'.' -f4)
+ NANO=$(echo 1.16.1 | cut -d'.' -f4)
if test x"$NANO" = x || test "x$NANO" = "x0" ; then
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: 0 (release)" >&5
@@ -8112,10 +8112,10 @@
done
- GST_CURRENT=1600
+ GST_CURRENT=1601
GST_REVISION=0
- GST_AGE=1600
- GST_LIBVERSION=1600:0:1600
+ GST_AGE=1601
+ GST_LIBVERSION=1601:0:1601
@@ -12749,10 +12749,10 @@
-GST_REQ=1.16.0
-GSTPB_REQ=1.16.0
-GSTPG_REQ=1.16.0
-GSTPD_REQ=1.16.0
+GST_REQ=1.16.1
+GSTPB_REQ=1.16.1
+GSTPG_REQ=1.16.1
+GSTPD_REQ=1.16.1
@@ -18953,7 +18953,7 @@
# report actual input values of CONFIG_FILES etc. instead of their
# values after options handling.
ac_log="
-This file was extended by GStreamer RTSP Server Library $as_me 1.16.0, which
was
+This file was extended by GStreamer RTSP Server Library $as_me 1.16.1, which
was
generated by GNU Autoconf 2.69. Invocation command line was
CONFIG_FILES = $CONFIG_FILES
@@ -19019,7 +19019,7 @@
cat >>$CONFIG_STATUS <<_ACEOF || ac_write_fail=1
ac_cs_config="`$as_echo "$ac_configure_args" | sed 's/^ //;
s/[\\""\`\$]/\\\\&/g'`"
ac_cs_version="\\
-GStreamer RTSP Server Library config.status 1.16.0
+GStreamer RTSP Server Library config.status 1.16.1
configured by $0, generated by GNU Autoconf 2.69,
with options \\"\$ac_cs_config\\"
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/configure.ac
new/gst-rtsp-server-1.16.1/configure.ac
--- old/gst-rtsp-server-1.16.0/configure.ac 2019-04-19 01:34:12.000000000
+0200
+++ new/gst-rtsp-server-1.16.1/configure.ac 2019-09-23 12:16:58.000000000
+0200
@@ -2,7 +2,7 @@
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], [1.16.0],
+AC_INIT([GStreamer RTSP Server Library], [1.16.1],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server])
AG_GST_INIT
@@ -53,13 +53,13 @@
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 1600, 0, 1600)
+AS_LIBTOOL(GST, 1601, 0, 1601)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.16.0
-GSTPB_REQ=1.16.0
-GSTPG_REQ=1.16.0
-GSTPD_REQ=1.16.0
+GST_REQ=1.16.1
+GSTPB_REQ=1.16.1
+GSTPG_REQ=1.16.1
+GSTPD_REQ=1.16.1
dnl *** autotools stuff ****
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPAuth.html
new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPAuth.html
--- old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPAuth.html 2019-04-19
01:34:57.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPAuth.html 2019-09-23
12:17:44.000000000 +0200
@@ -414,7 +414,6 @@
<p>Sets the certificate database that is used to verify peer certificates.
If set to <a
href="/usr/share/gtk-doc/html/glib/glib-Standard-Macros.html#NULL:CAPS"><code
class="literal">NULL</code></a> (the default), then peer certificate validation
will always
set the <a
href="/usr/share/gtk-doc/html/gio/gio-TLS-Overview.html#G-TLS-CERTIFICATE-UNKNOWN-CA:CAPS"><code
class="literal">G_TLS_CERTIFICATE_UNKNOWN_CA</code></a> error.</p>
-<p>Since 1.6</p>
<div class="refsect3">
<a name="gst-rtsp-auth-set-tls-database.parameters"></a><h4>Parameters</h4>
<div class="informaltable"><table class="informaltable" width="100%"
border="0">
@@ -437,6 +436,7 @@
</tbody>
</table></div>
</div>
+<p class="since">Since: 1.6</p>
</div>
<hr>
<div class="refsect2">
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPClient.html
new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPClient.html
--- old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPClient.html
2019-04-19 01:34:57.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPClient.html
2019-09-23 12:17:44.000000000 +0200
@@ -1392,7 +1392,7 @@
<tr>
<td class="struct_member_name"><p><em class="structfield"><code><a
name="GstRTSPClientClass.tunnel-http-response"></a>tunnel_http_response</code></em>
()</p></td>
<td class="struct_member_description"><p>called when a response to the GET
request is about to
-be sent for a tunneled connection. The response can be modified. Since
1.4</p></td>
+be sent for a tunneled connection. The response can be modified. Since:
1.4</p></td>
<td class="struct_member_annotations"> </td>
</tr>
<tr>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPStream.html
new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPStream.html
--- old/gst-rtsp-server-1.16.0/docs/libs/html/GstRTSPStream.html
2019-04-19 01:34:57.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/docs/libs/html/GstRTSPStream.html
2019-09-23 12:17:44.000000000 +0200
@@ -2216,6 +2216,7 @@
<p> <a
href="/usr/share/gtk-doc/html/glib/glib-Standard-Macros.html#TRUE:CAPS"><code
class="literal">TRUE</code></a> if <em
class="parameter"><code>stream</code></em>
is seekable, else <a
href="/usr/share/gtk-doc/html/glib/glib-Standard-Macros.html#FALSE:CAPS"><code
class="literal">FALSE</code></a>.</p>
</div>
+<p class="since">Since: 1.14</p>
</div>
<hr>
<div class="refsect2">
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/docs/libs/html/gst-rtsp-server-1.0.devhelp2
new/gst-rtsp-server-1.16.1/docs/libs/html/gst-rtsp-server-1.0.devhelp2
--- old/gst-rtsp-server-1.16.0/docs/libs/html/gst-rtsp-server-1.0.devhelp2
2019-04-19 01:34:57.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/docs/libs/html/gst-rtsp-server-1.0.devhelp2
2019-09-23 12:17:44.000000000 +0200
@@ -305,7 +305,7 @@
<keyword type="function" name="gst_rtsp_stream_set_pt_map ()"
link="GstRTSPStream.html#gst-rtsp-stream-set-pt-map"/>
<keyword type="function" name="gst_rtsp_stream_request_aux_sender ()"
link="GstRTSPStream.html#gst-rtsp-stream-request-aux-sender" since="1.6"/>
<keyword type="function" name="gst_rtsp_stream_request_aux_receiver ()"
link="GstRTSPStream.html#gst-rtsp-stream-request-aux-receiver" since="1.16"/>
- <keyword type="function" name="gst_rtsp_stream_seekable ()"
link="GstRTSPStream.html#gst-rtsp-stream-seekable"/>
+ <keyword type="function" name="gst_rtsp_stream_seekable ()"
link="GstRTSPStream.html#gst-rtsp-stream-seekable" since="1.14"/>
<keyword type="function" name="GstRTSPStreamTransportFilterFunc ()"
link="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc"/>
<keyword type="function" name="gst_rtsp_stream_transport_filter ()"
link="GstRTSPStream.html#gst-rtsp-stream-transport-filter"/>
<keyword type="function" name="gst_rtsp_stream_set_ulpfec_pt ()"
link="GstRTSPStream.html#gst-rtsp-stream-set-ulpfec-pt" since="1.16"/>
@@ -432,7 +432,7 @@
<keyword type="function" name="gst_rtsp_auth_get_tls_certificate ()"
link="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate"/>
<keyword type="function" name="gst_rtsp_auth_set_tls_certificate ()"
link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate"/>
<keyword type="function" name="gst_rtsp_auth_get_tls_database ()"
link="GstRTSPAuth.html#gst-rtsp-auth-get-tls-database" since="1.6"/>
- <keyword type="function" name="gst_rtsp_auth_set_tls_database ()"
link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-database"/>
+ <keyword type="function" name="gst_rtsp_auth_set_tls_database ()"
link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-database" since="1.6"/>
<keyword type="function" name="gst_rtsp_auth_get_tls_authentication_mode
()" link="GstRTSPAuth.html#gst-rtsp-auth-get-tls-authentication-mode"/>
<keyword type="function" name="gst_rtsp_auth_set_tls_authentication_mode
()" link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-authentication-mode"
since="1.6"/>
<keyword type="function" name="gst_rtsp_auth_set_realm ()"
link="GstRTSPAuth.html#gst-rtsp-auth-set-realm" since="1.16"/>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/docs/libs/html/index.html
new/gst-rtsp-server-1.16.1/docs/libs/html/index.html
--- old/gst-rtsp-server-1.16.0/docs/libs/html/index.html 2019-04-19
01:34:57.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/docs/libs/html/index.html 2019-09-23
12:17:44.000000000 +0200
@@ -15,7 +15,7 @@
<div>
<div><table class="navigation" id="top" width="100%" cellpadding="2"
cellspacing="0"><tr><th valign="middle"><p class="title">GStreamer RTSP Server
Reference Manual</p></th></tr></table></div>
<div><p class="releaseinfo">
- for GStreamer RTSP Server 1.16.0
+ for GStreamer RTSP Server 1.16.1
</p></div>
</div>
<hr>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-auth.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-auth.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-auth.c 2018-09-24
10:26:34.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-auth.c 2019-09-21
19:58:44.000000000 +0200
@@ -323,7 +323,7 @@
* If set to %NULL (the default), then peer certificate validation will always
* set the %G_TLS_CERTIFICATE_UNKNOWN_CA error.
*
- * Since 1.6
+ * Since: 1.6
*/
void
gst_rtsp_auth_set_tls_database (GstRTSPAuth * auth, GTlsDatabase * database)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-client.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-client.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-client.c 2019-02-18
17:11:58.000000000 +0100
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-client.c 2019-09-21
19:58:44.000000000 +0200
@@ -1830,7 +1830,8 @@
}
/* grab RTPInfo from the media now */
- rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
+ if (!(rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia)))
+ goto rtp_info_error;
/* construct the response now */
code = GST_RTSP_STS_OK;
@@ -1838,9 +1839,7 @@
gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
- if (rtpinfo)
- gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
- rtpinfo);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, rtpinfo);
if (seek_style)
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
seek_style);
@@ -1927,6 +1926,12 @@
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
return FALSE;
}
+rtp_info_error:
+ {
+ GST_ERROR ("client %p: failed to add RTP-Info", client);
+ send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
+ return FALSE;
+ }
}
static void
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-client.h
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-client.h
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-client.h 2019-02-18
17:11:58.000000000 +0100
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-client.h 2019-09-21
19:58:44.000000000 +0200
@@ -108,7 +108,7 @@
* @params_get: get parameters. This function should also initialize the
* RTSP response(ctx->response) via a call to
gst_rtsp_message_init_response()
* @tunnel_http_response: called when a response to the GET request is about to
- * be sent for a tunneled connection. The response can be modified. Since 1.4
+ * be sent for a tunneled connection. The response can be modified. Since:
1.4
*
* The client class structure.
*/
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-media-factory.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-media-factory.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-media-factory.c
2018-10-09 00:30:50.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-media-factory.c
2019-09-21 19:58:44.000000000 +0200
@@ -1511,6 +1511,8 @@
* Set the maximum time-to-live value of outgoing multicast packets.
*
* Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_factory_set_max_mcast_ttl (GstRTSPMediaFactory * factory,
@@ -1541,6 +1543,8 @@
* Get the the maximum time-to-live value of outgoing multicast packets.
*
* Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
*/
guint
gst_rtsp_media_factory_get_max_mcast_ttl (GstRTSPMediaFactory * factory)
@@ -1566,6 +1570,8 @@
*
* Decide whether the multicast socket should be bound to a multicast address
or
* INADDR_ANY.
+ *
+ * Since: 1.16
*/
void
gst_rtsp_media_factory_set_bind_mcast_address (GstRTSPMediaFactory * factory,
@@ -1589,6 +1595,8 @@
* Check if multicast sockets are configured to be bound to multicast
addresses.
*
* Returns: %TRUE if multicast sockets are configured to be bound to multicast
addresses.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_factory_is_bind_mcast_address (GstRTSPMediaFactory * factory)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-media.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-media.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-media.c 2019-04-10
01:10:51.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-media.c 2019-09-21
19:58:44.000000000 +0200
@@ -1869,6 +1869,8 @@
* Set the maximum time-to-live value of outgoing multicast packets.
*
* Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
@@ -1907,6 +1909,8 @@
* Get the the maximum time-to-live value of outgoing multicast packets.
*
* Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Since: 1.16
*/
guint
gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
@@ -1932,6 +1936,8 @@
*
* Decide whether the multicast socket should be bound to a multicast address
or
* INADDR_ANY.
+ *
+ * Since: 1.16
*/
void
gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
@@ -1960,6 +1966,8 @@
* Check if multicast sockets are configured to be bound to multicast
addresses.
*
* Returns: %TRUE if multicast sockets are configured to be bound to multicast
addresses.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
@@ -2637,6 +2645,8 @@
* the pipeline must contain all needed transport parts (transport sinks).
*
* Returns: %TRUE on success.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
@@ -4546,7 +4556,7 @@
}
/**
- * gst_rtsp_media_get_seekable:
+ * gst_rtsp_media_seekable:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media seek and up to what point in time,
@@ -4555,6 +4565,8 @@
* Returns: -1 if the stream is not seekable, 0 if seekable only to the
beginning
* and > 0 to indicate the longest duration between any two random access
points.
* %G_MAXINT64 means any value is possible.
+ *
+ * Since: 1.14
*/
GstClockTimeDiff
gst_rtsp_media_seekable (GstRTSPMedia * media)
@@ -4584,6 +4596,8 @@
* SETUP.
*
* Returns: %TRUE if the media pipeline has been sucessfully updated.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-onvif-media-factory.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-onvif-media-factory.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-onvif-media-factory.c
2018-03-13 14:19:19.000000000 +0100
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-onvif-media-factory.c
2019-09-21 19:58:44.000000000 +0200
@@ -330,9 +330,7 @@
g_mutex_clear (&factory->priv->lock);
- return
- G_OBJECT_CLASS (gst_rtsp_onvif_media_factory_parent_class)->finalize
- (object);
+ G_OBJECT_CLASS (gst_rtsp_onvif_media_factory_parent_class)->finalize
(object);
}
static void
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-onvif-media.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-onvif-media.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-onvif-media.c
2018-10-09 00:30:50.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-onvif-media.c
2019-09-21 19:58:44.000000000 +0200
@@ -212,7 +212,7 @@
g_mutex_clear (&media->priv->lock);
- return G_OBJECT_CLASS (gst_rtsp_onvif_media_parent_class)->finalize (object);
+ G_OBJECT_CLASS (gst_rtsp_onvif_media_parent_class)->finalize (object);
}
static void
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-sdp.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-sdp.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-sdp.c 2018-09-24
10:28:21.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-sdp.c 2019-09-21
19:58:44.000000000 +0200
@@ -184,6 +184,20 @@
}
}
+/**
+ * gst_rtsp_sdp_make_media:
+ * @sdp: a #GstRTSPMessage
+ * @info: a #GstSDPInfo
+ * @stream: a #GstRTSPStream
+ * @caps: a #GstCaps
+ * @profile: a #GstRTSPProfile
+ *
+ * Creates a #GstSDPMedia from the parameters and stores it in @sdp.
+ *
+ * Returns: %TRUE on success
+ *
+ * Since: 1.14
+ */
gboolean
gst_rtsp_sdp_make_media (GstSDPMessage * sdp, GstSDPInfo * info,
GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-session-media.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-session-media.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-session-media.c
2018-09-24 10:28:38.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-session-media.c
2019-09-21 19:58:44.000000000 +0200
@@ -416,6 +416,8 @@
*
* Returns: (transfer full) (element-type GstRTSPStreamTransport): a
* list of #GstRTSPStreamTransport, g_ptr_array_unref () after usage.
+ *
+ * Since: 1.14
*/
GPtrArray *
gst_rtsp_session_media_get_transports (GstRTSPSessionMedia * media)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-stream-transport.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-stream-transport.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-stream-transport.c
2019-02-18 17:11:58.000000000 +0100
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-stream-transport.c
2019-09-21 19:58:44.000000000 +0200
@@ -682,6 +682,8 @@
* @trans: a #GstRTSPStreamTransport
*
* Signal the installed message_sent callback for @trans.
+ *
+ * Since: 1.16
*/
void
gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport * trans)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-stream.c
new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-stream.c
--- old/gst-rtsp-server-1.16.0/gst/rtsp-server/rtsp-stream.c 2019-04-19
01:34:12.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst/rtsp-server/rtsp-stream.c 2019-09-21
19:58:44.000000000 +0200
@@ -2107,6 +2107,7 @@
*
* Returns: %TRUE if the requested ttl has been set successfully.
*
+ * Since: 1.16
*/
gboolean
gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream * stream, guint ttl)
@@ -2133,6 +2134,7 @@
*
* Returns: the maximum time-to-live value of outgoing multicast packets.
*
+ * Since: 1.16
*/
guint
gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream * stream)
@@ -2155,6 +2157,7 @@
*
* Returns: TRUE if the requested ttl value is allowed.
*
+ * Since: 1.16
*/
gboolean
gst_rtsp_stream_verify_mcast_ttl (GstRTSPStream * stream, guint ttl)
@@ -2176,6 +2179,8 @@
*
* Decide whether the multicast socket should be bound to a multicast address
or
* INADDR_ANY.
+ *
+ * Since: 1.16
*/
void
gst_rtsp_stream_set_bind_mcast_address (GstRTSPStream * stream,
@@ -2195,6 +2200,8 @@
* Check if multicast sockets are configured to be bound to multicast
addresses.
*
* Returns: %TRUE if multicast sockets are configured to be bound to multicast
addresses.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
@@ -4639,6 +4646,7 @@
* Get the multicast RTP socket from @stream for a @family.
*
* Returns: (transfer full) (nullable): the multicast RTP socket or %NULL if no
+ *
* socket could be allocated for @family. Unref after usage
*/
GSocket *
@@ -4674,6 +4682,8 @@
*
* Returns: (transfer full) (nullable): the multicast RTCP socket or %NULL if
no
* socket could be allocated for @family. Unref after usage
+ *
+ * Since: 1.14
*/
GSocket *
gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream * stream,
@@ -4712,6 +4722,8 @@
* allocated.
*
* Returns: %TRUE if @destination can be addedd and handled by @stream.
+ *
+ * Since: 1.16
*/
gboolean
gst_rtsp_stream_add_multicast_client_address (GstRTSPStream * stream,
@@ -4760,6 +4772,8 @@
* Get all multicast client addresses that RTP data will be sent to
*
* Returns: A comma separated list of host:port pairs with destinations
+ *
+ * Since: 1.16
*/
gchar *
gst_rtsp_stream_get_multicast_client_addresses (GstRTSPStream * stream)
@@ -5021,6 +5035,8 @@
* Unblocks the dataflow on @stream if it is linked.
*
* Returns: %TRUE on success
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_unblock_linked (GstRTSPStream * stream)
@@ -5237,6 +5253,8 @@
* Checks whether the individual @stream is seekable.
*
* Returns: %TRUE if @stream is seekable, else %FALSE.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_seekable (GstRTSPStream * stream)
@@ -5291,6 +5309,8 @@
* SETUP.
*
* Returns: %TRUE if the stream has been sucessfully updated.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_complete_stream (GstRTSPStream * stream,
@@ -5341,6 +5361,8 @@
* seek operations on it.
*
* Returns: %TRUE if the stream contains at least one sink element.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_is_complete (GstRTSPStream * stream)
@@ -5365,6 +5387,8 @@
* Checks whether the stream is a sender.
*
* Returns: %TRUE if the stream is a sender and %FALSE otherwise.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_is_sender (GstRTSPStream * stream)
@@ -5389,6 +5413,8 @@
* Checks whether the stream is a receiver.
*
* Returns: %TRUE if the stream is a receiver and %FALSE otherwise.
+ *
+ * Since: 1.14
*/
gboolean
gst_rtsp_stream_is_receiver (GstRTSPStream * stream)
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/gst-rtsp-server.doap
new/gst-rtsp-server-1.16.1/gst-rtsp-server.doap
--- old/gst-rtsp-server-1.16.0/gst-rtsp-server.doap 2019-04-19
01:34:12.000000000 +0200
+++ new/gst-rtsp-server-1.16.1/gst-rtsp-server.doap 2019-09-23
12:16:58.000000000 +0200
@@ -32,6 +32,16 @@
<release>
<Version>
+ <revision>1.16.1</revision>
+ <branch>1.16</branch>
+ <name></name>
+ <created>2019-09-23</created>
+ <file-release
rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.16.1.tar.xz";
/>
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.16.0</revision>
<branch>master</branch>
<name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.16.0/meson.build
new/gst-rtsp-server-1.16.1/meson.build
--- old/gst-rtsp-server-1.16.0/meson.build 2019-04-19 01:34:12.000000000
+0200
+++ new/gst-rtsp-server-1.16.1/meson.build 2019-09-23 12:16:58.000000000
+0200
@@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
- version : '1.16.0',
+ version : '1.16.1',
meson_version : '>= 0.47',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
