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Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 76 |
1 files changed, 76 insertions, 0 deletions
@@ -1,10 +1,86 @@ +=== release 1.18.1 === + +2020-10-26 11:15:28 +0000 Tim-Philipp Müller <tim@centricular.com> + + * ChangeLog: + * NEWS: + * RELEASE: + * gst-rtsp-server.doap: + * meson.build: + Release 1.18.1 + +2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: make use of blocked_running_time in query_position + When blocking, the sink element will not have received a buffer + yet and the position query will fail. Instead, we make use of + the running time of the buffer we blocked on. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/163> + +2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: collect rtp info when blocking + We don't unblock the stream anymore before replying to the + play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443), + so the sinks don't have a last-sample after potentially flush + seeking. seek_trickmode waits for preroll however, which means + the stream will block and wait for a first buffer. Subsequent + calls to get_rtpinfo() can thus make use of the information. + See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115 + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/163> + +2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-server-internal.h: + * gst/rtsp-server/rtsp-stream.c: + rtsp-media: set a 0 storage size for TCP receivers + ulpfec correction is obviously useless when receiving a stream + over TCP, and in TCP modes the rtp storage receives non + timestamped buffers, causing it to queue buffers indefinitely, + until the queue grows so large that sanity checks kick in and + warnings start to get emitted. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/158> + +2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: preroll on gap events + This allows negotiating a SDP with all streams present, but only + start sending packets at some later point in time + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/157> + +2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: do not unblock on unsuspend + rtsp_media_unsuspend() is called from handle_play_request() + before sending the play response. Unblocking the streams here + was causing data to be sent out before the client was ready + to handle it, with obvious side effects such as initial packets + getting discarded, causing decoding errors. + Instead we can simply let the media streams be unblocked when + the state of the media is set to PLAYING, which occurs after + sending the play response. + Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/156> + +2020-09-08 17:44:37 +0100 Tim-Philipp Müller <tim@centricular.com> + + * docs/gst_plugins_cache.json: + * meson.build: + Back to development + === release 1.18.0 === 2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com> + * .gitlab-ci.yml: * ChangeLog: * NEWS: * RELEASE: + * docs/gst_plugins_cache.json: * gst-rtsp-server.doap: * meson.build: Release 1.18.0 |