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authorTim-Philipp Müller <tim@centricular.com>2019-02-26 11:58:53 +0000
committerTim-Philipp Müller <tim@centricular.com>2019-02-26 11:58:53 +0000
commit14d0b77df67c0beff0a4095ecacceaedd28143dc (patch)
tree81abf721cb025931b2c7255a493839130fc7e2d5 /ChangeLog
parent7e01dfd151c35868ac679ca12f30caaefa9bbb8b (diff)
Release 1.15.21.15.2
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+=== release 1.15.2 ===
+
+2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.2
+
+2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/client.c:
+ rtsp-media: Fix multicast use case with common media
+ Use case
+ client 1: SETUP
+ client 1: PLAY
+ client 2: SETUP
+ client 1: TEARDOWN
+ client 2: PLAY
+ client 2: TEARDOWN
+
+2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: remove recursive behavior
+ Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
+
+2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Only allow to set either a send_func or send_messages_func but not both
+ And route all messages through the send_func if no send_messages_func
+ was provided.
+ We otherwise break backwards compatibility.
+
+2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-client: Add support for sending buffer lists directly
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtsp-server: Add support for buffer lists
+ This adds new functions for passing buffer lists through the different
+ layers without breaking API/ABI, and enables the appsink to actually
+ provide buffer lists.
+ This should already reduce CPU usage and potentially context switches a
+ bit by passing a whole buffer list from the appsink instead of
+ individual buffers. As a next step it would be necessary to
+ a) Add support for a vector of data for the GstRTSPMessage body
+ b) Add support for sending multiple messages at once to the
+ GstRTSPWatch and let it be handled internally
+ c) Adding API to GOutputStream that works like writev()
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix crash in close handler
+ The close handler could trigger a crash because it invalidated the
+ watch_context while still leaving a source attached to it which would be
+ cleaned up at a later point.
+
+2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Use cached address when allocating sockets
+ If an address/port was previously decided upon (ex: multicast in the
+ SDP), then use that instead of re-creating another one
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
+
+2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix race codition in finish_unprepare
+ The previous fix for race condition around finish_unprepare where the
+ function could be called twice assumed that the status wouldn't change
+ during execution of the function. This assumption is incorrect as the
+ state may change, for example if an error message arrives from the
+ pipeline bus.
+ Instead a flag keeping track on whether the finish_unprepare function
+ is currently executing is introduced and checked.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
+
=== release 1.15.1 ===
2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>