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author | Tim-Philipp Müller <tim@centricular.com> | 2019-02-26 11:58:53 +0000 |
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committer | Tim-Philipp Müller <tim@centricular.com> | 2019-02-26 11:58:53 +0000 |
commit | 14d0b77df67c0beff0a4095ecacceaedd28143dc (patch) | |
tree | 81abf721cb025931b2c7255a493839130fc7e2d5 /ChangeLog | |
parent | 7e01dfd151c35868ac679ca12f30caaefa9bbb8b (diff) |
Release 1.15.21.15.2
Diffstat (limited to 'ChangeLog')
-rw-r--r-- | ChangeLog | 101 |
1 files changed, 101 insertions, 0 deletions
@@ -1,3 +1,104 @@ +=== 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> |