diff options
| author | Tim-Philipp Müller <tim@centricular.com> | 2019-01-17 02:26:48 +0000 |
|---|---|---|
| committer | Tim-Philipp Müller <tim@centricular.com> | 2019-01-17 02:26:50 +0000 |
| commit | d5ccb5a7d0fc76460c646811a558c2304ac03f1f (patch) | |
| tree | 24ad66440240966ef577364c5ea9068f055b9447 /ChangeLog | |
| parent | 3be1b9bba8a165a62008871eff3e9cfcb1338910 (diff) | |
Release 1.15.1gst-rtsp-server-1.15.1
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 1037 |
1 files changed, 1037 insertions, 0 deletions
@@ -1,3 +1,1040 @@ +=== release 1.15.1 === + +2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com> + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + * meson.build: + Release 1.15.1 + +2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + Add source elements to the pipeline before activation + In plug_src we changed the element state before adding it to + the owner container. This prevented the pipeline from intercepting + a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order + to assign a custom task pool. + Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53 + +2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com> + + * common: + Automatic update of common submodule + From ed78bee to 59cb678 + +2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at> + + * examples/test-appsrc.c: + examples: test-appsrc: fix coding style error + +2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at> + + * examples/test-appsrc.c: + examples: test-appsrc: fix buffer leak + +2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Update priv->blocked when linked streams are unblocked. + Media is considered to be blocked when all streams that belong to + that media are blocked. + This patch solves the problem of inconsistent updates of + priv->blocked that are not synchronized with the media state. + +2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Don't block streams before seeking + Before the seek operation is performed on media, it's required that + its pipeline is prepared <=> the pipeline is in the PAUSED state. + At this stage, all transport parts (transport sinks) have been successfully + added to the pipeline and there is no need for blocking the streams. + +2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com> + + * tests/check/gst/rtspserver.c: + tests: rtspserver: Add shared media test case for TCP + +2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Use seqnum-offset for rtpinfo + The sequence number in the rtpinfo is supposed to be the first RTP + sequence number. The "seqnum" property on a payloader is supposed to be + the number from the last processed RTP packet. The sequence number for + payloaders that inherit gstrtpbasepayload will not be correct in case of + buffer lists. In order to fix the seqnum property on the payloaders + gst-rtsp-server must get the sequence number for rtpinfo elsewhere and + "seqnum-offset" from the "stats" property contains the value of the + very first RTP packet in a stream. The server will, however, try to look + at the last simple in the sink element and only use properties on the + payloader in case there no sink elements yet, and by looking at the last + sample of the sink gives the server full control of which RTP packet it + looks at. If the payloader does not have the "stats" property, "seqnum" + is still used since "seqnum-offset" is only present in as part of + "stats" and this is still an issue not solved with this patch. + Needed for gst-plugins-base!17 + +2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Plug memory leak + Attaching a GSource to a context will increase the refcount. The idle + source will never be free'd since the initial reference is never + dropped. + +2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com> + + * .gitlab-ci.yml: + Add Gitlab CI configuration + This commit adds a .gitlab-ci.yml file, which uses a feature + to fetch the config from a centralized repository. The intent is + to have all the gstreamer modules use the same configuration. + The configuration is currently hosted at the gst-ci repository + under the gitlab/ci_template.yml path. + Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29 + +2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com> + + * .gitmodules: + * gst-rtsp-server.doap: + Update git locations to gitlab + +2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/meson.build: + meson: add new onvif types + +2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/meson.build: + Add ONVIF subclass headers to the installed headers in meson.build too + +2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-server-object.h: + * gst/rtsp-server/rtsp-server.h: + rtsp-server: Declare GstRTSPServer struct before anything else + It's needed by all kinds of other headers, including the ones that are + required for defining the GstRTSPServer struct itself and its API. + +2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-onvif-client.h: + * gst/rtsp-server/rtsp-onvif-media-factory.h: + * gst/rtsp-server/rtsp-onvif-media.h: + * gst/rtsp-server/rtsp-onvif-server.h: + Mark all ONVIF-specific subclasses as Since 1.14 + +2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/meson.build: + * gst/rtsp-server/rtsp-context.h: + * gst/rtsp-server/rtsp-onvif-server.c: + * gst/rtsp-server/rtsp-onvif-server.h: + * gst/rtsp-server/rtsp-server-object.h: + * gst/rtsp-server/rtsp-server-prelude.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session.h: + Include ONVIF types from single-include rtsp-server.h + ... by actually making it a single-include header and moving everything + related to the GstRTSPServer type to rtsp-server-object.h instead. + Otherwise there are too many circular includes. + https://bugzilla.gnome.org/show_bug.cgi?id=797361 + +2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-latency-bin.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-stream: use idle source in on_message_sent + When the underlying layers are running on_message_sent, this sometimes + causes the underlying layer to send more data, which will cause the + underlying layer to run callback on_message_sent again. This can go on + and on. + To break this chain, we introduce an idle source that takes care of + sending data if there are more to send when running callback + https://bugzilla.gnome.org/show_bug.cgi?id=797289 + +2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Remove timeout GSource on cleanup + Avoids ending up with races where a timeout would still be around + *after* a client was gone. This could happen rather easily in + RTSP-over-HTTP mode on a local connection, where each RTSP message + would be sent as a different HTTP connection with the same tunnelid. + If not properly removed, that timeout would then try to free again + a client (and its contents). + +2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/Makefile.am: + autotools: fix distcheck + +2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/meson.build: + * gst/rtsp-server/rtsp-latency-bin.c: + * gst/rtsp-server/rtsp-latency-bin.h: + * gst/rtsp-server/rtsp-onvif-media.c: + onvif: encapsulate onvif part into a bin + ...and thus do not let onvif affect pipelines latency + https://bugzilla.gnome.org/show_bug.cgi?id=797174 + +2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com> + + * tests/check/gst/client.c: + tests: client: Avoid bind() failures in tests + https://bugzilla.gnome.org/show_bug.cgi?id=797059 + +2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/mediafactory.c: + New property for socket binding to mcast addresses + By default the multicast sockets are bound to INADDR_ANY, + as it's not allowed to bind sockets to multicast addresses + in Windows. This default behaviour can be changed by setting + bind-mcast-address property on the media-factory object. + https://bugzilla.gnome.org/show_bug.cgi?id=797059 + +2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/meson.build: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-context.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-server-prelude.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-token.c: + * meson.build: + libs: fix API export/import and 'inconsistent linkage' on MSVC + Export rtsp-server library API in headers when we're building the + library itself, otherwise import the API from the headers. + This fixes linker warnings on Windows when building with MSVC. + Fix up some missing config.h includes when building the lib which + is needed to get the export api define from config.h + https://bugzilla.gnome.org/show_bug.cgi?id=797185 + +2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/rtsp-media-factory.c: + rtsp-media-factory: Add missing break statements + This resulted in warnings/assertions whenever one accessed the + max-mcast-ttl property. + CID #1439515 + CID #1439523 + +2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com> + + * meson.build: + * meson_options.txt: + meson: add gobject-cast-checks, glib-asserts, glib-checks options + +2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/meson.build: + * meson_options.txt: + * tests/check/meson.build: + meson: add option to disable build of rtspclientsink plugin + +2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com> + + * meson_options.txt: + meson: re-arrange options + +2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * meson.build: + * meson_options.txt: + * tests/check/meson.build: + * tests/meson.build: + meson: Use feature option for tests option + This was somehow missed the last time around. + +2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * gst/rtsp-server/meson.build: + * meson.build: + meson: Maintain macOS ABI through dylib versioning + Requires Meson 0.48, but the feature will be ignored on older versions + so it's safe to add it without bumping the requirement. + Documentation: + https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library + +2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com> + + * gst/rtsp-sink/meson.build: + * meson.build: + meson: add pkg-config file for the rtspclientsink plugin + +2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/client.c: + rtsp-client: Avoid reuse of channel numbers for interleaved + If a (strange) client would reuse interleaved channel numbers in + multiple SETUP requests, we should not accept them. The channel + numbers are used for looking up stream transports in the + priv->transports hash table, and transports disappear from the table + if channel numbers are reused. + RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the + server to change the channel numbers suggested by the client. + https://bugzilla.gnome.org/show_bug.cgi?id=796988 + +2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com> + + * tests/check/gst/client.c: + rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP + Allow regex for matching transport header against expected pattern. + https://bugzilla.gnome.org/show_bug.cgi?id=796988 + +2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * tests/check/meson.build: + meson: There is no gstreamer-plugins-good-1.0.pc + There is no installed version of that, only an uninstalled version. + +2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/stream.c: + Fix indentation again + +2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/stream.c: + stream: Added a list of multicast client addresses + When media is shared, the same media stream can be sent + to multiple multicast groups. Currently, there is no API + to retrieve multicast addresses from the stream. + When calling gst_rtsp_stream_get_multicast_address() function, + only the first multicast address is returned. + With this patch, each multicast destination requested in SETUP + will be stored in an internal list (call to + gst_rtsp_stream_add_multicast_client_address()). + The list of multicast groups requested by the clients can be + retrieved by calling gst_rtsp_stream_get_multicast_client_addresses(). + There still exist some problems with the current implementation + in the multicast case: + 1) The receiving part is currently only configured with + regard to the first multicast client (see + https://bugzilla.gnome.org/show_bug.cgi?id=796917). + 2) Secondly, of security reasons, some constraints should be + put on the requested multicast destinations (see + https://bugzilla.gnome.org/show_bug.cgi?id=796916). + Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + stream: Choose the maximum ttl value provided by multicast clients + The maximum ttl value provided so far by the multicast clients + will be chosen and reported in the response to the current + client request. + Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com> + + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/client.c: + rtsp-stream: Don't require address pool in the transport specific case + If "transport.client-settings" parameter is set to true, the client is + allowed to specify destination, ports and ttl. + There is no need for pre-configured address pool. + Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1 + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/client.c: + client: Don't reserve multicast address in the client setting case + When two multicast clients request specific transport + configurations, and "transport.client-settings" parameter is + set to true, it's wrong to actually require that these two + clients request the same multicast group. + Removed test_client_multicast_invalid_transport_specific test + cases as they wrongly require that the requested destination + address is supposed to be present in the address pool, also in + the case when "transport.client-settings" parameter is set to true. + Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/mediafactory.c: + Add new API for setting/getting maximum multicast ttl value + Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233 + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: avoid duplicating the first multicast client + In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so + clients were dynamically added and removed to the multicast + udp sinks, as such we should no longer add a first client in + set_multicast_socket_for_udpsink + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + Revert "rtsp-stream: avoid duplicating the first multicast client" + This reverts commit 33570944401747f44d8ebfec535350651413fb92. + Commits where accidentially squashed together + +2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/mediafactory.c: + Revert "Add new API for setting/getting maximum multicast ttl value" + This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52. + Commits where accidentially squashed together + +2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/client.c: + Revert "rtsp-stream: Don't require address pool in the transport specific case" + This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52. + Commits where accidentially squashed together + +2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/stream.c: + Revert "stream: Choose the maximum ttl value provided by multicast clients" + This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0. + Commits where accidentially squashed together + +2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com> + + * examples/test-auth-digest.c: + examples: Fix indentation + +2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/stream.c: + stream: Choose the maximum ttl value provided by multicast clients + The maximum ttl value provided so far by the multicast clients + will be chosen and reported in the response to the current + client request. + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com> + + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/client.c: + rtsp-stream: Don't require address pool in the transport specific case + If "transport.client-settings" parameter is set to true, the client is + allowed to specify destination, ports and ttl. + There is no need for pre-configured address pool. + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/client.c: + * tests/check/gst/mediafactory.c: + Add new API for setting/getting maximum multicast ttl value + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: avoid duplicating the first multicast client + In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so + clients were dynamically added and removed to the multicast + udp sinks, as such we should no longer add a first client in + set_multicast_socket_for_udpsink + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com> + + * gst/rtsp-server/Makefile.am: + rtsp-server: Add gstreamer-base gir dir in autotools + +2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-client: always allocate both IPV4 and IPV6 sockets + multiudpsink does not support setting the socket* properties + after it has started, which meant that rtsp-server could no + longer serve on both IPV4 and IPV6 sockets since the patches + from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were + merged. + When first connecting an IPV6 client then an IPV4 client, + multiudpsink fell back to using the IPV6 socket. + When first connecting an IPV4 client, then an IPV6 client, + multiudpsink errored out, released the IPV4 socket, then + crashed when trying to send a message on NULL nevertheless, + that is however a separate issue. + This could probably be fixed by handling the setting of + sockets in multiudpsink after it has started, that will + however be a much more significant effort. + For now, this commit simply partially reverts the behaviour + of rtsp-stream: it will continue to only create the udpsinks + when needed, as was the case since the patches were merged, + it will however when creating them, always allocate both + sockets and set them on the sink before it starts, as was + the case prior to the patches. + Transport configuration will only error out if the allocation + of UDP sockets fails for the actual client's family, this + also downgrades the GST_ERRORs in alloc_ports_one_family + to GST_WARNINGs, as failing to allocate is no longer + necessarily fatal. + https://bugzilla.gnome.org/show_bug.cgi?id=796875 + +2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * meson.build: + * meson_options.txt: + meson: Convert common options to feature options + These are necessary for gst-build to set options correctly. The + remaining automagic option is cgroup support in examples. + https://bugzilla.gnome.org/show_bug.cgi?id=795107 + +2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Slightly simplify locking + +2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + Limit queued TCP data messages to one per stream + Before, the watch backlog size in GstRTSPClient was changed + dynamically between unlimited and a fixed size, trying to avoid both + unlimited memory usage and deadlocks while waiting for place in the + queue. (Some of the deadlocks were described in a long comment in + handle_request().) + In the previous commit, we changed to a fixed backlog size of 100. + This is possible, because we now handle RTP/RTCP data messages differently + from RTSP request/response messages. + The data messages are messages tunneled over TCP. We allow at most one + queued data message per stream in GstRTSPClient at a time, and + successfully sent data messages are acked by sending a "message-sent" + callback from the GstStreamTransport. Until that ack comes, the + GstRTSPStream does not call pull_sample() on its appsink, and + therefore the streaming thread in the pipeline will not be blocked + inside GstRTSPClient, waiting for a place in the queue. + pull_sample() is called when we have both an ack and a "new-sample" + signal from the appsink. Then, we know there is a buffer to write. + RTSP request/response messages are not acked in the same way as data + messages. The rest of the 100 places in the queue are used for + them. If the queue becomes full of request/response messages, we + return an error and close the connection to the client. + Change-Id: I275310bc90a219ceb2473c098261acc78be84c97 + +2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Use fixed backlog size + Change to using a fixed backlog size WATCH_BACKLOG_SIZE. + Preparation for the next commit, which changes to a different way of + avoiding both deadlocks and unlimited memory usage with the watch + backlog. + +2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: unref clock (if set) when finalizing + https://bugzilla.gnome.org/show_bug.cgi?id=796814 + +2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org> + + * docs/libs/gst-rtsp-server-sections.txt: + rtsp-media: add gst_rtsp_media_*_set_clock to docs + https://bugzilla.gnome.org/show_bug.cgi?id=796814 + +2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: unref old clock when setting new clock + https://bugzilla.gnome.org/show_bug.cgi?id=796724 + +2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com> + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: unref clock in finalize + https://bugzilla.gnome.org/show_bug.cgi?id=796724 + +2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-onvif-media.c: + rtsp-onvif-media: fix g-ir-scanner warnings + +2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + .gitignore: add another example binary + +2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com> + + * examples/meson.build: + meson: add new test-appsrc2 example to meson build + +2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com> + + * examples/Makefile.am: + examples: fix build of new test-appsrc2 example + Need to link against libgstapp-1.0. + +2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com> + + * examples/.gitignore: + * examples/Makefile.am: + * examples/test-appsrc2.c: + examples: Add test-appsrc2 + Add an example of feeding both audio and video into an RTSP + pipeline via appsrc. + +2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com> + + * gst/rtsp-server/rtsp-client.c: + client: Strip transport parts as whitespaces could be around commas + https://bugzilla.gnome.org/show_bug.cgi?id=758428 + +2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup + Fix race when setting up source elements. + Since we set the source element(s) to PLAYING state before hooking + them up to the downstream funnel, it's possible for the source element + to receive packets before we actually get to linking it to the funnel, + in which case buffers would be pushed out on an unlinked pad, causing + it to error out and stop receiving more data. + We fix this by blocking the source's srcpad until we have linked it. + https://bugzilla.gnome.org/show_bug.cgi?id=796160 + +2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Fix mismatch between allowed and configured protocols + https://bugzilla.gnome.org/show_bug.cgi?id=796679 + +2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Emit a signal when the SRTP decoder is created + https://bugzilla.gnome.org/show_bug.cgi?id=778080 + +2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Don't require presence of sinks in _get_*_socket() + Transport specific sink elements are added to the pipeline + in PLAY request and sockets are already created in SETUP so + it's actually wrong to require the presence of sinks in + _get_*_socket() functions. + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Update transport for multicast clients as well + If a multicast client requests different transport settings + than the existing one make sure that this new transport + configuruation is propagated to the multicast udp sink. + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Set the multicast TTL parameter on multicast udp sinks + And not on unicast udp sinks + https://bugzilla.gnome.org/show_bug.cgi?id=793441 + +2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-thread-pool.c: + Update for g_type_class_add_private() deprecation in recent GLib + +2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-stream.c: + Fix indentation + +2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com> + + * examples/Makefile.am: + * examples/test-video-disconnect.c: + examples: Add test-video-disconnect example + Simple example which cuts off all clients 10 seconds + after the first one connects. + +2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * docs/libs/gst-rtsp-server-sections.txt: + * examples/test-auth-digest.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + rtsp-auth: Add support for parsing .htdigest files + Passwords are usually not stored in clear text, but instead + stored already hashed in a .htdigest file. + Add support for parsing such files, add API to allow setting + a custom realm in RTSPAuth, and update the digest example. + https://bugzilla.gnome.org/show_bug.cgi?id=796637 + +2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + * gst/rtsp-sink/gstrtspclientsink.h: + rtspclientsink: fix waiting for multiple streams + We were previously only ever waiting for a single stream to notify it's + blocked status through GstRTSPStreamBlocking. Actually count streams to + wait for. + Fixes rtspclientsink sending SDP's without out some of the input + streams. + https://bugzilla.gnome.org/show_bug.cgi?id=796624 + +2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * docs/libs/gst-rtsp-server-sections.txt: + docs: add missing auth methods + +2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: only create funnel if it didn't exist already. + This precented using multiple protocols for the same stream. + https://bugzilla.gnome.org/show_bug.cgi?id=796634 + +2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * examples/meson.build: + meson: build auth-digest example + +2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + Get payloader stats only for the sending streams + Get/set payloader properties only for streams that actually + contain a payloader element. + https://bugzilla.gnome.org/show_bug.cgi?id=796523 + +2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/Makefile.am: + Makefile: Don't hardcode libtool for g-i build + Similar to the other commits in core/base/bad + +2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com> + + * gst/rtsp-server/rtsp-onvif-media-factory.h: + rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel + https://bugzilla.gnome.org/show_bug.cgi?id=796229 + +2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Don't deadlock in preroll on early close + If the connection is closed very early, the flushing + marker might not get set and rtspclientsink can get + deadlocked waiting for preroll forever. + https://bugzilla.gnome.org/show_bug.cgi?id=786961 + +2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * meson.build: + * meson_options.txt: + meson: Update option names to omit disable_ and with- prefixes + Also yield common options to the outer project (gst-build in our case) + so that they don't have to be set manually. + +2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com> + + * meson.build: + meson: use -Wl,-Bsymbolic-functions where supported + Just like the autotools build. + +2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + * tests/check/Makefile.am: + configure: check for -good and -bad plugins only in uninstalled setup + Avoids confusing configure messages looking or a -good .pc file + that doesn't exist. + Also use plugindir variables that common macros set while at it. + https://bugzilla.gnome.org/show_bug.cgi?id=795466 + +2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Fix session timeout + When streaming data over TCP then is not the keep-alive + functionality working. + The reason is that the function do_send_data have changed + to boolean but the code is still checking the received result + from send_func with GST_RTSP_OK. + The result is that a successful send_func will always lead to + that do_send_data is returning false and the keep-alive will + not be updated. + https://bugzilla.gnome.org/show_bug.cgi?id=795321 + +2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * gst/rtsp-sink/gstrtspclientsink.c: + * gst/rtsp-sink/gstrtspclientsink.h: + Implement support for ULP Forward Error Correction + In this initial commit, interface is only exposed for RECORD, + further work will be needed in rtspsrc to support this for + PLAY. + https://bugzilla.gnome.org/show_bug.cgi?id=794911 + +2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-onvif-media.c: + Revert "rtsp-server: Switch around sendonly/recvonly attributes" + This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57. + While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of + the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say + the opposite, just like the ONVIF standard. + Let's follow those RFCs as we're doing RTSP here, and add a property at + a later time if needed to switch to the SDP RFC behaviour. + https://bugzilla.gnome.org/show_bug.cgi?id=793964 + +2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com> + + * common: + Automatic update of common submodule + From 3fa2c9e to ed78bee + +2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/rtspclientsink.c: + gst: Run everything through gst-indent again + +2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + rtsp-media: query the position on active streams if media is complete + If the media is complete, i.e. one or more streams have been configured + with sinks, then we want to query the position on those streams only. + A query on an incomplete stream may return a position that originates from + an earlier preroll. + https://bugzilla.gnome.org/show_bug.cgi?id=794964 + +2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: make sure not to use freed string + Set transport string to NULL after freeing it, so that + at worst we get a NULL pointer if constructing a new + transport string fails (which shouldn't really fail here). + Also check return value of that, just in case. + CID 1433768. + +2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: do not free string passed to take_header + +2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: do not take lock in request_aux_receiver + Added it right before pushing the previous commit, it is + incorrect and deadlocks because this function gets called + from the join_bin thread, which already holds the lock, + that's the reason why request_aux_sender didn't take the + lock either. + +2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-server: add API to enable retransmission requests + "do-retransmission" was previously set when rtx-time != 0, + which made no sense as do-retransmission is used to enable + the sending of retransmission requests, where as rtx-time + is used by the peer to enable storing of buffers in order + to respond to retransmission requests. + rtsp-media now also provides a callback for the + request-aux-receiver signal. + https://bugzilla.gnome.org/show_bug.cgi?id=794822 + +2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: add rtx ssrc to mikey's crypto sessions + https://bugzilla.gnome.org/show_bug.cgi?id=794813 + +2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response + This in order to be able to decrypt the RTCP backchannel + https://bugzilla.gnome.org/show_bug.cgi?id=794813 + +2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Send KeyMgmt header in ANNOUNCE response + When sending back an encrypted RTCP back channel, it is useful + for the client to know the encryption key. + https://bugzilla.gnome.org/show_bug.cgi?id=794813 + +2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-stream: extract handle_keymgmt from rtsp-client + rtspclientsink will also need to parse KeyMgmt headers + sent by the server to decrypt the RTCP backchannel stream + https://bugzilla.gnome.org/show_bug.cgi?id=794813 + +2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + * tests/check/gst/rtspclientsink.c: + rtspclientsink: Fix client ports for the RTCP backchannel + This was broken since the work for delayed transport creation + was merged: the creation of the transports string depends on + calling stream_get_server_port, which only starts returning + something meaningful after a call to stream_allocate_udp_sockets + has been made, this function expects a transport that we parse + from the transport string ... + Significant refactoring is in order, but does not look entirely + trivial, for now we put a band aid on and create a second transport + string after the stream has been completed, to pass it in + the request headers instead of the previous, incomplete one. + https://bugzilla.gnome.org/show_bug.cgi?id=794789 + +2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client:Error handling when equal http session cookie + There are some clients that are sending same session cookie on random + basis. + https://bugzilla.gnome.org/show_bug.cgi?id=753616 + +2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + rtsp-media-factory-uri: Fix compilation with latest GLib + rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’: + rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types] + data->factory = g_object_ref (factory); + ^ + +2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com> + + * NEWS: + * RELEASE: + * configure.ac: + * meson.build: + Back to development + === release 1.14.0 === 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com> |