2016-11-01 16:06:46 +00:00
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=== release 1.10.0 ===
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2016-11-01 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.10.0
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2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/rtspserver.c:
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* tests/check/gst/stream.c:
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tests: try to avoid using the same ports in different tests
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Causes problems with client multicast tests otherwise if
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tests are run in parallel.
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https://bugzilla.gnome.org/show_bug.cgi?id=773640
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2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/client.c:
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tests: client: use fail_unless_equals_foo() for better failure reporting
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2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: Session filter in unwatch session
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Call session filter with filter_session_media as paramer in
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client_unwatch_session if using drop_backlog = FALSE.
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In client_unwatch_session its allowed to grow the watchs backlog.
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If using drop_backlog = FALSE and the backlog is full it will cause
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a deadlock when setting session media state to NULL
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if the backlog is not allowed to grow.
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https://bugzilla.gnome.org/show_bug.cgi?id=771983
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2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
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* meson.build:
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meson: add fallbacks for gst modules
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For gst-all.
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2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: Fix factory leaking in find_media() in error cases
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https://bugzilla.gnome.org/show_bug.cgi?id=771488
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2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: Fix randomly missing streams from SDP with dynamic elements
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When using dynamic elements, gst_rtsp_stream_join_bin() is called from
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"pad-added" signal. In that case priv->srcpad could already have its caps,
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and they'll be sent to priv->send_src[0] pad. That means that when it
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connects "notify::caps" signal, that pad could already have received its
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caps and the signal won't be emitted anymore.
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In that case priv->caps stay to NULL and when building the SDP that stream
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gets ignored. Leading to missing video or audio when playing in client side.
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https://bugzilla.gnome.org/show_bug.cgi?id=772478
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2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
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* meson.build:
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meson: update version
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2016-09-30 10:04:12 +00:00
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=== release 1.9.90 ===
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2016-11-01 16:06:46 +00:00
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2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
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2016-09-30 10:04:12 +00:00
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2016-11-01 16:06:46 +00:00
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* ChangeLog:
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* NEWS:
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* RELEASE:
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2016-09-30 10:04:12 +00:00
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* configure.ac:
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2016-11-01 16:06:46 +00:00
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* gst-rtsp-server.doap:
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Release 1.9.90
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2016-09-30 10:04:12 +00:00
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2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-server: Hint that set_multicast_iface expects the name of the interface
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To prevent any possibly confusion with IPs or anything else.
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https://bugzilla.gnome.org/show_bug.cgi?id=771530
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2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
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https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
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2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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configure: Depend on gstreamer 1.9.2.1
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2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
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* autogen.sh:
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* common:
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Automatic update of common submodule
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From b18d820 to f980fd9
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2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
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* autogen.sh:
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* common:
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Automatic update of common submodule
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From 6f2d209 to b18d820
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2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Remove unused _locked() variant of a function
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It was added during refactoring.
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2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: cosmetic cleanup
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: Compare IP addresses case insensitive in more places
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* common:
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* gst/rtsp-server/rtsp-stream.c:
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stream: Fix leaked joined_bin
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There is no need to keep a strong ref on it, and _leave_bin() was
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setting it to NULL before calling g_clear_object() so it was leaked.
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Compare IP address strings case insensitive
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Otherwise IPv6 addresses might fail this comparision.
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2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Bind multicast sockets to ANY as before
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https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
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2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
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* gst/rtsp-server/rtsp-session.c:
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rtsp-session: Fix segfault when doing keep-alive after removing the session
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If keep-alive happens after removing the session but before finalizing the
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stream transport, we would segfault.
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https://bugzilla.gnome.org/show_bug.cgi?id=750544
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2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Always create multicast UDP elements if the protocol flag is set
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Adding them later will cause deadlocks due to
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1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
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2) adding the multicast sink
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3) waiting for it to get data to preroll again
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3) never happens because the queues after the tee are full.
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2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Fix up various multicast related issues
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2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
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* tests/check/gst/stream.c:
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tests: Fix compilation
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2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-stream.c:
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* tests/check/gst/stream.c:
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stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
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This is basically reverting changes introduced in commit f62a9a7,
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because it was introducing various regressions:
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- It introduces a leak of udpsrc elements that got wrongly fixed by adding
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an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
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ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
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- If a mcast client connects, it creates a new socket in SETUP to try to respect
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the destination/port given by the client in the transport, and overrides the
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socket already set on the udpsink element. That means that if we already had a
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client connected, the source address on the udp packets it receives suddenly
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changes.
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- If a 2nd mcast client connects, the destination/port in its transport is
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ignored but its transport wasn't updated.
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What this patch does:
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- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
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- Always have a tee+queue when udp is enabled. This could be optimized
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again in a later patch, but is more complicated. If no unicast clients
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connects then those elements are useless, this could be also optimized
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in a later patch.
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- When mcast transport is added, it creates a new set of udpsrc/udpsink,
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seperated from those for unicast clients. Since we already support only
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one mcast address, we also create only one set of elements.
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: factor our plug_src function
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: factor out plug_sink function
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: small documentation clarification
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: Keep a ref on joined bin
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: code cleanup
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: small fix in error code path
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
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This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
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but keeps unit tests.
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https://bugzilla.gnome.org/show_bug.cgi?id=766612
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2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
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|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2016-09-01 09:32:51 +00:00
|
|
|
|
=== release 1.9.2 ===
|
|
|
|
|
|
|
2016-09-30 10:04:12 +00:00
|
|
|
|
2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
|
2016-09-01 09:32:51 +00:00
|
|
|
|
|
2016-09-30 10:04:12 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-09-01 09:32:51 +00:00
|
|
|
|
* configure.ac:
|
2016-09-30 10:04:12 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.9.2
|
2016-09-01 09:32:51 +00:00
|
|
|
|
|
|
|
|
|
|
2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* config.h.meson:
|
|
|
|
|
|
* examples/meson.build:
|
|
|
|
|
|
* gst/meson.build:
|
|
|
|
|
|
* gst/rtsp-server/meson.build:
|
|
|
|
|
|
* gst/rtsp-sink/meson.build:
|
|
|
|
|
|
* meson.build:
|
|
|
|
|
|
* pkgconfig/meson.build:
|
|
|
|
|
|
* tests/check/meson.build:
|
|
|
|
|
|
* tests/meson.build:
|
|
|
|
|
|
Add support for Meson as alternative/parallel build system
|
|
|
|
|
|
https://github.com/mesonbuild/meson
|
|
|
|
|
|
|
|
|
|
|
|
2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
build: silence error about pthread for 'make check' in osx
|
|
|
|
|
|
Fixes "clang: error: argument unused during compilation: '-pthread'"
|
|
|
|
|
|
|
|
|
|
|
|
2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Fix leaking of media in error cases
|
|
|
|
|
|
With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
|
|
|
|
|
|
and myself to make the media refcounting a bit easier to follow.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=755632
|
|
|
|
|
|
|
|
|
|
|
|
2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Fix leaking of session in error cases
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=755632
|
|
|
|
|
|
|
|
|
|
|
|
2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f363b32 to f49c55e
|
|
|
|
|
|
|
|
|
|
|
|
2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2016-07-06 10:28:12 +00:00
|
|
|
|
=== release 1.9.1 ===
|
|
|
|
|
|
|
2016-09-01 09:32:51 +00:00
|
|
|
|
2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
|
2016-07-06 10:28:12 +00:00
|
|
|
|
|
2016-09-01 09:32:51 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-07-06 10:28:12 +00:00
|
|
|
|
* configure.ac:
|
2016-09-01 09:32:51 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.9.1
|
2016-07-06 10:28:12 +00:00
|
|
|
|
|
|
|
|
|
|
2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: Need to add -DGST_STATIC_COMPILATION when building only statically
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=767463
|
|
|
|
|
|
|
|
|
|
|
|
2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From ac2f647 to f363b32
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
sdp: add rollover counters for all sender SSRC
|
|
|
|
|
|
We add different crypto sessions in MIKEY, one for each sender
|
|
|
|
|
|
SSRC. Currently, all of them will have the same security policy, 0.
|
|
|
|
|
|
The rollover counters are obtained from the srtpenc element using the
|
|
|
|
|
|
"stats" property.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=730539
|
|
|
|
|
|
|
|
|
|
|
|
2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
docs: fix some typos
|
|
|
|
|
|
|
|
|
|
|
|
2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
g-i: pass compiler env to g-ir-scanner
|
|
|
|
|
|
It's what introspection.mak does as well. Should
|
|
|
|
|
|
fix spurious build failures on gnome-continuous
|
|
|
|
|
|
(caused by g-ir-scanner getting compiler details
|
|
|
|
|
|
via python which is broken in some environments
|
|
|
|
|
|
so passing the compiler details bypasses that).
|
|
|
|
|
|
|
|
|
|
|
|
2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
|
|
|
|
|
|
This works with rtspsrc and live555, but fails with e.g. ffmpeg.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=766619
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: Check return value of sscanf
|
|
|
|
|
|
And just make sure we always have 0/0 if we have an error
|
|
|
|
|
|
CID #1352031
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
* tests/check/gst/stream.c:
|
|
|
|
|
|
rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
|
|
|
|
|
|
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
|
|
|
|
|
|
- Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
|
|
|
|
|
|
- Create unit test for shared media.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=764744
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
|
|
|
|
|
|
For IPv6 addresses, binding to a multicast group does not work on Linux
|
|
|
|
|
|
either. Always bind to ANY and then later join the multicast group.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=764679
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6f2d209 to ac2f647
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
rtsp-thread-pool: explained why GSource is a part of ThreadImpl
|
|
|
|
|
|
Clarified why it is necessary to add source information to
|
|
|
|
|
|
GstRTSPThreadImpl. See the reported bug in GLib:
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=720186
|
|
|
|
|
|
for more information.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=761702
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
examples: Clean up CFLAGS/LDADD even more
|
|
|
|
|
|
The internal .la should come first and is part of LDADD, as is
|
|
|
|
|
|
GST_CFLAGS/LIBS.
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
|
|
|
|
|
|
|
|
|
|
|
|
2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 18:39:05 +0200 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-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-server: Implement clock signalling according to RFC7273
|
|
|
|
|
|
For NTP and PTP clocks we signal the actual clock that is used and signal
|
|
|
|
|
|
the direct media clock offset.
|
|
|
|
|
|
For all other clocks we at least signal that it's the local sender clock.
|
|
|
|
|
|
This allows receivers to know which clock was used to generate the media and
|
|
|
|
|
|
its RTP timestamps. Receivers can then implement network synchronization,
|
|
|
|
|
|
either absolute or at least relative by getting the sender clock rate directly
|
|
|
|
|
|
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
|
|
|
|
|
|
times.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=760005
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: Add support for setting the multicast interface
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=763000
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.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:
|
|
|
|
|
|
rtsp-media: Add support for setting the multicast interface
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=763000
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: use new gst_element_class_add_static_pad_template()
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=763196
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2016-03-24 11:00:35 +00:00
|
|
|
|
=== release 1.8.0 ===
|
|
|
|
|
|
|
2016-07-06 10:28:12 +00:00
|
|
|
|
2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2016-03-24 11:00:35 +00:00
|
|
|
|
|
2016-07-06 10:28:12 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-03-24 11:00:35 +00:00
|
|
|
|
* configure.ac:
|
2016-07-06 10:28:12 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.8.0
|
2016-03-24 11:00:35 +00:00
|
|
|
|
|
|
|
|
|
|
2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
|
|
|
|
|
|
This would get us NO_PREROLL in the bin again and break seeking.
|
|
|
|
|
|
Thanks to Carlos Rafael Giani for helping to debug this!
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=740509
|
|
|
|
|
|
|
2016-03-15 10:26:13 +00:00
|
|
|
|
=== release 1.7.91 ===
|
|
|
|
|
|
|
2016-03-24 11:00:35 +00:00
|
|
|
|
2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2016-03-15 10:26:13 +00:00
|
|
|
|
|
2016-03-24 11:00:35 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-03-15 10:26:13 +00:00
|
|
|
|
* configure.ac:
|
2016-03-24 11:00:35 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.7.91
|
2016-03-15 10:26:13 +00:00
|
|
|
|
|
|
|
|
|
|
2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
|
|
|
|
|
|
Without this, RECORD pipelines are broken because
|
|
|
|
|
|
a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
|
|
|
|
|
|
added later. Previously it was there earlier and due to NO_PREROLL caused the
|
|
|
|
|
|
pipeline to preroll immediately
|
|
|
|
|
|
b) the udpsrc for the pipeline is added later and never set to PLAYING state,
|
|
|
|
|
|
as the corresponding code previously was only for PLAY pipelines.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=763281
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Fix typo in the docstring
|
|
|
|
|
|
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Disable multicast loopback for all our sockets
|
|
|
|
|
|
On Windows this is a receiver-side setting, on Linux a sender-side setting. As
|
|
|
|
|
|
we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
|
|
|
|
|
|
loopback setting on the socket... while udpsink does which unfortunately has
|
|
|
|
|
|
no effect here on Windows but on Linux.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/stream.c:
|
|
|
|
|
|
stream tests: added new tests
|
|
|
|
|
|
Test a case when the address pool only contains multicast addresses
|
|
|
|
|
|
and the client is requesting unicast udp.
|
|
|
|
|
|
Added tests for multicast ports allocation.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Only bind multicast sockets to ANY on Windows
|
|
|
|
|
|
On Linux it is still needed to bind to the multicast address
|
|
|
|
|
|
to filter out random other packets, while on Windows binding
|
|
|
|
|
|
to multicast addresses just fails.
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
|
|
|
|
|
|
Otherwise we fail to allocate UDP ports if the pool only contains multicast
|
|
|
|
|
|
addresses, which is something that used to work before. For unicast addresses
|
|
|
|
|
|
if the pool contains none, we just allocate them as if there is no pool at
|
|
|
|
|
|
all.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-server: Fix indentation
|
|
|
|
|
|
|
|
|
|
|
|
2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Don't bind the sockets to multicast addresses
|
|
|
|
|
|
This works on Linux but fails completely on Windows. You're supposed
|
|
|
|
|
|
to bind to ANY and then join the multicast group.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
2016-03-01 17:00:45 +00:00
|
|
|
|
=== release 1.7.90 ===
|
|
|
|
|
|
|
2016-03-15 10:26:13 +00:00
|
|
|
|
2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2016-03-01 17:00:45 +00:00
|
|
|
|
|
2016-03-15 10:26:13 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-03-01 17:00:45 +00:00
|
|
|
|
* configure.ac:
|
2016-03-15 10:26:13 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.7.90
|
2016-03-01 17:00:45 +00:00
|
|
|
|
|
|
|
|
|
|
2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From b64f03f to 6f2d209
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
* tests/check/gst/rtspclientsink.c:
|
|
|
|
|
|
rtspsink: Fix some leaks in rtspclientsink and the unit test.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=762525
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
* tests/check/gst/rtspclientsink.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
* tests/check/gst/stream.c:
|
|
|
|
|
|
tests: unit test fixes
|
|
|
|
|
|
Removed port allocation test from the media suite.
|
|
|
|
|
|
The port allocation failure is now in the stream suite.
|
|
|
|
|
|
rtspserver:
|
|
|
|
|
|
Make sure that the media is suspended after the DESCRIBE request
|
|
|
|
|
|
before reconfiguring the UDP sinks.
|
|
|
|
|
|
rtspclientsink:
|
|
|
|
|
|
In the RECORD case we have to set async property to false
|
|
|
|
|
|
for the appsink element in the test in order to make sure
|
|
|
|
|
|
that the media pipeline doesn't hang in start_preroll().
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-stream: postpone UDP socket allocation until SETUP
|
|
|
|
|
|
Postpone the allocation of the UDP sockets until we know
|
|
|
|
|
|
what transport has been chosen by the client.
|
|
|
|
|
|
Both unicast and multicast UDP sources are created in one
|
|
|
|
|
|
function.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: postpone the creation of the UDP sources
|
|
|
|
|
|
Code refactoring: allocate the UDP ports after the sender and
|
|
|
|
|
|
the reciver parts have been created.
|
|
|
|
|
|
We postpone the creation of the UDP sources until the UDP
|
|
|
|
|
|
ports have been allocated.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: added function for setting UDP sources to PLAYING state
|
|
|
|
|
|
Code refactoring: Introduced a function for setting UDP sources
|
|
|
|
|
|
to PLAYING state.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: added function for creating and configuring UDP sources
|
|
|
|
|
|
Code refactoring: create and configure UDP sources in a separate function.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: added function for RTP/RTCP socket configuration
|
|
|
|
|
|
Code refactoring: configure RTP and RTCP sockets for UDP sinks
|
|
|
|
|
|
in a separate function.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: added function for creating and configuring UDP sinks
|
|
|
|
|
|
Code refactoring: create and configure UDP sinks in a separate function.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: added helper function for creating the sender/receiver parts
|
|
|
|
|
|
Code refactoring: introduced helper function for creating
|
|
|
|
|
|
the receiver and the sender parts of the streaming pipeline.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757488
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2016-02-19 10:03:18 +00:00
|
|
|
|
=== release 1.7.2 ===
|
|
|
|
|
|
|
2016-03-01 17:00:45 +00:00
|
|
|
|
2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2016-02-19 10:03:18 +00:00
|
|
|
|
|
2016-03-01 17:00:45 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2016-02-19 10:03:18 +00:00
|
|
|
|
* configure.ac:
|
2016-03-01 17:00:45 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.7.2
|
2016-02-19 10:03:18 +00:00
|
|
|
|
|
|
|
|
|
|
2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
|
|
|
|
|
|
uninstalled.pc: add support for non libtool build systems
|
|
|
|
|
|
Currently the .la path is provided which requires to use libtool as
|
|
|
|
|
|
mentioned in the GStreamer manual section-helloworld-compilerun.html.
|
|
|
|
|
|
It is fine as long as the application is built using libtool.
|
|
|
|
|
|
So currently it is not possible to compile a GStreamer application
|
|
|
|
|
|
within gst-uninstalled with CMake or other build system different
|
|
|
|
|
|
than autotools.
|
|
|
|
|
|
This patch allows to do the following in gst-uninstalled env:
|
|
|
|
|
|
gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
|
|
|
|
|
|
gstreamer-rtsp-server-1.0)
|
|
|
|
|
|
Previously it required to prepend libtool --mode=link
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=720778
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: remove check for impossible condition
|
|
|
|
|
|
Goto error label checks stream to see if it needs to be unreferenced before
|
|
|
|
|
|
returning, but this goto jumps happens before the stream is ever set, so it
|
|
|
|
|
|
will always be NULL in this error label.
|
|
|
|
|
|
CID #1352034
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: clean switch statements
|
|
|
|
|
|
Coverity demands for fallthrough statements to be clearly commented,
|
|
|
|
|
|
to distinguish from accidental fall throughs. And it also needs all
|
|
|
|
|
|
cases to finish with a break, even if the break is never going to be
|
|
|
|
|
|
executed like in the case of a continue jump.
|
|
|
|
|
|
CID #1352039
|
|
|
|
|
|
CID #1352040
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: extend the AM_TESTS_ENVIRONMENT from check.mak
|
|
|
|
|
|
To get the CK_DEFAULT_TIMEOUT defined for all tests
|
|
|
|
|
|
Also removes a 120 seconds timeout that was set as default
|
|
|
|
|
|
explicitly in this module
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=761472
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 86e4663 to b64f03f
|
|
|
|
|
|
|
|
|
|
|
|
2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: fix state_lock not locked again when preroll fails
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=761399
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: Move plugin specific flags below all the others
|
|
|
|
|
|
They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
|
|
|
|
|
|
-no-undefined. And -no-undefined is required on Windows to build DLLs.
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
rtspclientsink: Simplify slightly using new -base API
|
|
|
|
|
|
Use the new Mikey and SDP API in the base plugins libs
|
|
|
|
|
|
to simplify some code.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758180
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-sink/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.c:
|
|
|
|
|
|
* gst/rtsp-sink/gstrtspclientsink.h:
|
|
|
|
|
|
* gst/rtsp-sink/plugin.c:
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/rtspclientsink.c:
|
|
|
|
|
|
rtspsink: Add rtspclientsink element
|
|
|
|
|
|
Add an rtspclientsink element that accepts streams for which
|
|
|
|
|
|
there is a registered payloader and sends them to
|
|
|
|
|
|
an RTSP server using RECORD.
|
|
|
|
|
|
Sending is synchronised to the pipeline clock. Payload-types
|
|
|
|
|
|
are automatically selected. The 'new-payloader' signal is fired
|
|
|
|
|
|
for custom configuration of payloaders when they are created.
|
|
|
|
|
|
Can now stream a movie like this:
|
|
|
|
|
|
receiver:
|
|
|
|
|
|
./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
|
|
|
|
|
|
decodebin name=depay1 ! audioconvert ! autoaudiosink )"
|
|
|
|
|
|
sender:
|
|
|
|
|
|
gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
|
|
|
|
|
|
queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758180
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-stream: Add functions for using rtsp-stream from the client
|
|
|
|
|
|
Add a boolean to indicate that the rtsp-stream is running on the
|
|
|
|
|
|
'client' side of an RTSP connection, for sending streams via
|
|
|
|
|
|
RECORD. In that case, the roles of the client/server ports
|
|
|
|
|
|
in transport setup are swapped.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758180
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
rtsp-sdp: Add gst_rtsp_sdp_from_stream()
|
|
|
|
|
|
A new function that adds info from a GstRTSPStream into an SDP message.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758180
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Fix mutex beeing unlocked while they should be locked
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=761226
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
rtsp-media-factory: add missing break in "clock" property setter
|
|
|
|
|
|
CID 1348453
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: fixed assert during update transport
|
|
|
|
|
|
When RTSP server trying update transport during multicast, it throws an
|
|
|
|
|
|
assert. The assert is thrown because it is trying to get the parent of
|
|
|
|
|
|
an non-existing funnel element.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=760150
|
|
|
|
|
|
|
|
|
|
|
|
2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.h:
|
|
|
|
|
|
docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
|
|
|
|
|
|
gtk-doc can handle static inline functions just fine these days,
|
|
|
|
|
|
there's no need for this stuff any more.
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: replace duplicated codes to call new base sdp apis
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=745880
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock.c:
|
|
|
|
|
|
test-netclock: Use the new API to configure a clock directly
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.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:
|
|
|
|
|
|
rtsp-media: Add API to directly configure a clock on the media pipelines
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
rtsp-media-factory: Add FIXME for 2.0
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Fix indentation
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Do not prepare media after media times out
|
|
|
|
|
|
Deferred calls to start_prepare() can be deferred past the point until
|
|
|
|
|
|
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
|
|
|
|
|
|
prepared to wait. Previously there was no lock and no check for this
|
|
|
|
|
|
situation. This meant that a media could be prepared and unprepared
|
|
|
|
|
|
simultaneously by two different threads. Now a lock is in place and a
|
|
|
|
|
|
suitable check is done.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-09 18:24:24 +0200 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:
|
|
|
|
|
|
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
|
|
|
|
|
|
Without TEARDOWN it might be desireable to keep the media running and continue
|
|
|
|
|
|
sending data to the client, even if the RTSP connection itself is
|
|
|
|
|
|
disconnected.
|
|
|
|
|
|
Only do this for session medias that have only UDP transports. If there's at
|
|
|
|
|
|
least on TCP transport, it will stop working and cause problems when the
|
|
|
|
|
|
connection is disconnected.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758999
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2015-12-24 13:54:06 +00:00
|
|
|
|
=== release 1.7.1 ===
|
|
|
|
|
|
|
2016-02-19 10:03:18 +00:00
|
|
|
|
2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
|
2015-12-24 13:54:06 +00:00
|
|
|
|
|
2016-02-19 10:03:18 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-12-24 13:54:06 +00:00
|
|
|
|
* configure.ac:
|
2016-02-19 10:03:18 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.7.1
|
2015-12-24 13:54:06 +00:00
|
|
|
|
|
|
|
|
|
|
2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: Make -Bsymbolic check work with clang.
|
|
|
|
|
|
Update the -Bsymbolic check with the version glib has. This version
|
|
|
|
|
|
works with clang.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=759713
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
rtsp-session-pool: Avoid dollar sign ($) in session ids
|
|
|
|
|
|
Live555 in VLC strips off dollar signs and then gets very confused,
|
|
|
|
|
|
we don't loose too much entropy by just skipping it.
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.h:
|
|
|
|
|
|
rtsp-server: Add g_autoptr() support to all types
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=754464
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: fixed valgrind error
|
|
|
|
|
|
Fixed the valgrind error in unit test. The UDP source created during
|
|
|
|
|
|
gst_rtsp_stream_join_bin() was not released while destroying the rtp
|
|
|
|
|
|
bin.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=759010
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From b319909 to 86e4663
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: suspend media during setup request
|
|
|
|
|
|
SETUP request from clients needs to suspend the media to clear the
|
|
|
|
|
|
prerolled buffers. Otherwise it will not affect the prerolled buffer
|
|
|
|
|
|
and the prerolled buffers will be incorrect (for example block-size
|
|
|
|
|
|
from setup request will not affect the prerolled buffer unless the
|
|
|
|
|
|
media is suspended).
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758268
|
|
|
|
|
|
|
|
|
|
|
|
2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: create stream pipeline based on transport
|
|
|
|
|
|
Based on the protocol, create the rtsp stream pipeline. If only TCP or
|
|
|
|
|
|
only UDP is set as the transport protocol, it will not add the extra tee
|
|
|
|
|
|
or queue element to the pipeline. Both these elements will be added, if
|
|
|
|
|
|
it supports both TCP and UDP protocols. This improves the pipeline
|
|
|
|
|
|
performance when one protocol is present.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758179
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
|
|
|
|
|
|
Adding them when not needed will start some logic inside rtpbin that might be
|
|
|
|
|
|
problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
|
|
|
|
|
|
would start up a rtpjitterbuffer and behave in weird ways.
|
|
|
|
|
|
We still set up the UDP sources for RTP receiving for a sender media to be
|
|
|
|
|
|
able to receive any packets sent by the client for NAT traversal. They will
|
|
|
|
|
|
all go to a fakesink though.
|
|
|
|
|
|
Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
|
|
|
|
|
|
NO_PREROLL, which will cause deadlocks when seeking the media as it will never
|
|
|
|
|
|
receive ASYNC_DONE after a seek.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=758319
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Disable multicast loopback for the multicast udp sources too
|
|
|
|
|
|
On POSIX this setting is for sender sockets, on Windows for receiver sockets.
|
|
|
|
|
|
Previously we were only setting this for sender sockets, which caused looped
|
|
|
|
|
|
back packets to be received on Windows if a multicast transport was used.
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record-auth.c:
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
examples: Actually use the provided port in the record examples
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record-auth.c:
|
|
|
|
|
|
test-record-auth: Add the option to build in TLS support
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
test-auth: Use an 'anonymous' user for unauthenticated default
|
|
|
|
|
|
There's a comment on one of the resources that 'user' and 'admin'
|
|
|
|
|
|
shouldn't even be able to see it, but they can if the default
|
|
|
|
|
|
token is 'admin2', since that gives them access anyway.
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-record-auth.c:
|
|
|
|
|
|
Add test-record-auth example
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
|
|
|
|
|
|
|
|
|
|
|
|
2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: Change the logic so we don't pop a NULL context
|
|
|
|
|
|
When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
|
|
|
|
|
|
will sometimes fail. This call is made before any context is pushed
|
|
|
|
|
|
resulting in an attempt to pop a NULL context.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=757949
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtspserver: Add udp-mcast transport SETUP test
|
|
|
|
|
|
Refactor utility functions in the test file so they can handle
|
|
|
|
|
|
more than UDP and TCP as lower transport.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=756969
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Always unref return value of gst_object_get_parent()
|
|
|
|
|
|
Fixes a leak of a GstBin in the udp-mcast case.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=756968
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From b99800a to b319909
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Use new GST_ENABLE_EXTRA_CHECKS #define
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=756870
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6babecd to b99800a
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Update GLib dependency to 2.40.0
|
|
|
|
|
|
|
|
|
|
|
|
2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: listen to sender ssrc signals
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=746747
|
|
|
|
|
|
|
|
|
|
|
|
2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
common: update for new suppression
|
|
|
|
|
|
Makes check-valgrind pass with glib 2.46
|
|
|
|
|
|
|
|
|
|
|
|
2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Take reference to media that will be prepared
|
|
|
|
|
|
default_prepare() takes a transfer-none reference GstRTSPMedia object.
|
|
|
|
|
|
Later on a g_idle_source_new() is created and a pointer to the media
|
|
|
|
|
|
object is passed as user data. If the media is freed before the idle
|
|
|
|
|
|
source is dispatched the media object pointer is invalid, but the idle
|
|
|
|
|
|
source callback expects it to still be valid. To fix this a reference to
|
|
|
|
|
|
the media object is taken when registering the source callback function
|
|
|
|
|
|
and a corresponding release of the reference is done when the souce is
|
|
|
|
|
|
destroyed.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
rtsp-server: Fix memory leaks when context parse fails
|
|
|
|
|
|
When g_option_context_parse fails, context and error variables are not getting free'd
|
|
|
|
|
|
which results in memory leaks. Free'ing the same.
|
|
|
|
|
|
And replacing g_error_free with g_clear_error, which checks if the error being passed
|
|
|
|
|
|
is not NULL and sets the variable to NULL on free'ing.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753863
|
|
|
|
|
|
|
|
|
|
|
|
2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2015-09-25 21:32:52 +00:00
|
|
|
|
=== release 1.6.0 ===
|
|
|
|
|
|
|
2015-12-24 13:54:06 +00:00
|
|
|
|
2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2015-09-25 21:32:52 +00:00
|
|
|
|
|
2015-12-24 13:54:06 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-09-25 21:32:52 +00:00
|
|
|
|
* configure.ac:
|
2015-12-24 13:54:06 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.6.0
|
2015-09-25 21:32:52 +00:00
|
|
|
|
|
2015-09-18 18:12:06 +00:00
|
|
|
|
=== release 1.5.91 ===
|
|
|
|
|
|
|
2015-09-25 21:32:52 +00:00
|
|
|
|
2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2015-09-18 18:12:06 +00:00
|
|
|
|
|
2015-09-25 21:32:52 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-09-18 18:12:06 +00:00
|
|
|
|
* configure.ac:
|
2015-09-25 21:32:52 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.5.91
|
2015-09-18 18:12:06 +00:00
|
|
|
|
|
|
|
|
|
|
2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: fix docs for recently-added get/set_buffer_size API
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=749095
|
|
|
|
|
|
|
|
|
|
|
|
2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Don't crash on encrypted RTX SDP
|
|
|
|
|
|
In parse_keymgmt(), don't mutate the input string that's been passed
|
|
|
|
|
|
as const, especially since we might need the original value again if
|
|
|
|
|
|
the same key info applies to multiple streams (RTX, for example).
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=754753
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
test-mp4: Support filenames with spaces in them. Error out on too few arguments
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
test-record: Check parameter count and print out help
|
|
|
|
|
|
If no launch pipeline was supplied, print out some help
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-stream: Implement UDP buffer size setting.
|
|
|
|
|
|
Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
|
|
|
|
|
|
UDP TX buffer size.
|
|
|
|
|
|
Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
rtsp-media: Fix small typo causing gtk-doc to complain
|
|
|
|
|
|
|
2015-08-19 11:15:23 +00:00
|
|
|
|
=== release 1.5.90 ===
|
|
|
|
|
|
|
2015-09-18 18:12:06 +00:00
|
|
|
|
2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
|
2015-08-19 11:15:23 +00:00
|
|
|
|
|
2015-09-18 18:12:06 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-08-19 11:15:23 +00:00
|
|
|
|
* configure.ac:
|
2015-09-18 18:12:06 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.5.90
|
2015-08-19 11:15:23 +00:00
|
|
|
|
|
|
|
|
|
|
2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: get port number through gst_rtsp_url_get_port
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753473
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media-test: Removing unnecessary assertion
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753385
|
|
|
|
|
|
|
|
|
|
|
|
2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
Document that source keeps a ref on server until it's destroyed
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=749227
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media-test: Test for multiple dynamic payload
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753385
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: Only add fakesink once per pipeline
|
|
|
|
|
|
The intention is to prevent going PLAYING state before pads are created.
|
|
|
|
|
|
If there was mutilple dynamic payload, it would leak few fakesink and
|
|
|
|
|
|
actually prevent from ever reaching playing state.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753385
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Revert "rtsp-media: Only add 1 fakesink per pipeline"
|
|
|
|
|
|
This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
|
|
|
|
|
|
|
|
|
|
|
|
2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Only add 1 fakesink per pipeline
|
|
|
|
|
|
There should be only one fakesink per pipeline, not per dynpay. This
|
|
|
|
|
|
would lead to element naming clash.
|
|
|
|
|
|
|
|
|
|
|
|
2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: assertion error due to wrong condition check
|
|
|
|
|
|
In media to caps function, reserved_keys array is being used for variable i,
|
|
|
|
|
|
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
|
|
|
|
|
|
changed it to variable j
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753009
|
|
|
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|
2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Strip keys from the fmtp that we use internally in our caps
|
|
|
|
|
|
Skip keys from the fmtp, which we already use ourselves for the
|
|
|
|
|
|
caps. Some software is adding random things like clock-rate into
|
|
|
|
|
|
the fmtp, and we would otherwise here set a string-typed clock-rate
|
|
|
|
|
|
in the caps... and thus fail to create valid RTP caps
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=753009
|
|
|
|
|
|
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|
|
2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=752640
|
|
|
|
|
|
|
|
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|
|
2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f74b2df to 9aed1d7
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2015-06-24 21:44:37 +00:00
|
|
|
|
=== release 1.5.2 ===
|
|
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|
|
|
|
2015-08-19 11:15:23 +00:00
|
|
|
|
2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2015-06-24 21:44:37 +00:00
|
|
|
|
|
2015-08-19 11:15:23 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-06-24 21:44:37 +00:00
|
|
|
|
* configure.ac:
|
2015-08-19 11:15:23 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.5.2
|
2015-06-24 21:44:37 +00:00
|
|
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|
|
|
|
|
|
|
2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
|
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|
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|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
rtsp-client: allow application to decide what requirements are supported
|
|
|
|
|
|
Add "check-requirements" signal and vfunc to allow application
|
|
|
|
|
|
(and subclasses) to check the requirements.
|
|
|
|
|
|
Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=749417
|
|
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|
|
2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6015d26 to f74b2df
|
|
|
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|
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|
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|
|
2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Always use real payloader when creating streams
|
|
|
|
|
|
A bin that contains the real payloader might be used as payloader. In this
|
|
|
|
|
|
case we have to get the real payloader for the various properties it provides.
|
|
|
|
|
|
Example use cases for this are bins that payload some media and then have
|
|
|
|
|
|
additional elements that add metadata or RTP extension headers to the stream.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=750800
|
|
|
|
|
|
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|
|
|
|
2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
* examples/test-netclock.c:
|
|
|
|
|
|
test-netclock: Use new ntp-time-source property on rtpbin
|
|
|
|
|
|
Select the clock time to be used as NTP time source. This allows proper
|
|
|
|
|
|
synchronization between receivers, independent of sharing base times, and just
|
|
|
|
|
|
requires them to use the same clock.
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
* examples/test-netclock.c:
|
|
|
|
|
|
test-netclock: Setting the same base time on sender and receiver is not necessary
|
|
|
|
|
|
It's going to be fixed up by rtpbin when using ntp-sync=TRUE
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=750764
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server.types:
|
|
|
|
|
|
docs: add missing types
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=750764
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
docs: add missing apis
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=750764
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
GstRTSPAuth: Add client certificate authentication support
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=750471
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
test-netclock-client: Use new GstClock API to wait for clock synchronization
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
test-netclock-client: Use a GMainLoop and playbin's source-setup signal
|
|
|
|
|
|
A mainloop is needed to get glimagesink to display something on OSX, and
|
|
|
|
|
|
the source-setup signal just makes things a little bit easier.
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From d9a3353 to 6015d26
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From d37af32 to d9a3353
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 21ba2e5 to d37af32
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From c408583 to 21ba2e5
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
docs: remove variables that we define in the snippet from common
|
|
|
|
|
|
This is syncing our Makefile.am with upstream gtkdoc.
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 44a3517 to c408583
|
|
|
|
|
|
|
|
|
|
|
|
2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2015-06-07 09:20:01 +00:00
|
|
|
|
=== release 1.5.1 ===
|
|
|
|
|
|
|
2015-06-24 21:44:37 +00:00
|
|
|
|
2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2015-06-07 09:20:01 +00:00
|
|
|
|
|
2015-06-24 21:44:37 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2015-06-07 09:20:01 +00:00
|
|
|
|
* configure.ac:
|
2015-06-24 21:44:37 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.5.1
|
2015-06-07 09:20:01 +00:00
|
|
|
|
|
|
|
|
|
|
2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: No flush during Teardown.
|
|
|
|
|
|
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
|
|
|
|
|
|
backlog is empty it can happen that just a part of a message will be
|
|
|
|
|
|
sent and rest is in backlog queue. If then flush during teardown
|
|
|
|
|
|
just a part of message will be sent.This can lead to client miss
|
|
|
|
|
|
teardown response since it expect to get the last part of message.
|
|
|
|
|
|
The flushing during teardown was introduced to fix a deadlock that now
|
|
|
|
|
|
is fixed more generally in handle_request by temporary setting backlog
|
|
|
|
|
|
size to unlimited.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: Use AM_TESTS_ENVIRONMENT
|
|
|
|
|
|
Needed by the new automake test runner and the
|
|
|
|
|
|
current version of the common submodule.
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-server: Use single-include rtsp header to make sure we get all definitions
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Mark some more functions static
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Only unblock the media in suspend() when actually changing the state
|
|
|
|
|
|
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video-rtx.c:
|
|
|
|
|
|
examples: Use AVPF profile for the RTX example
|
|
|
|
|
|
|
|
|
|
|
|
2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-sdp: Only add RTX to the SDP when using a feedback profile
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: get valid clock-rate from last-sample
|
|
|
|
|
|
clock-rate in last-sample's caps is integer, not unsigned.
|
|
|
|
|
|
To get this value properly, variable needs to be type-casted to int.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=747614
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
autogen.sh: only run autopoint if gettext requested in configure.ac
|
|
|
|
|
|
Not just because there happens to be a po directory.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=748058
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Revert "configure.ac: uncomment gettext version setup"
|
|
|
|
|
|
This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
|
|
|
|
|
|
We don't need a gettext setup here and there's no po
|
|
|
|
|
|
directory either, so no reason why autopoint would be
|
|
|
|
|
|
run in the first place.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=748058
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-multicast.c:
|
|
|
|
|
|
* examples/test-multicast2.c:
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
* examples/test-video-rtx.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* tests/test-cleanup.c:
|
|
|
|
|
|
* tests/test-reuse.c:
|
|
|
|
|
|
Fix timeout function signatures across tests and examples
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
|
|
|
|
|
|
Make sure the test environment is set up.
|
|
|
|
|
|
https://bugzilla.gnome.org//show_bug.cgi?id=747624
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump automake requirement to 1.14 and autoconf to 2.69
|
|
|
|
|
|
This is only required for builds from git, people can still
|
|
|
|
|
|
build tarballs if they only have older autotools.
|
|
|
|
|
|
https://bugzilla.gnome.org//show_bug.cgi?id=747624
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure.ac: uncomment gettext version setup
|
|
|
|
|
|
Fixes autogen.sh. It would run autopoint, which would complain
|
|
|
|
|
|
that it could not find the gettext version in configure.ac.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=748058
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video-rtx.c:
|
|
|
|
|
|
test-video-rtx: set exact payload type to PCMA payloader
|
|
|
|
|
|
Setting wrong payload type causes failure to do retransmission through audio stream
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=747839
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-stream: fix to get valid each stream data for request-aux-sender signal
|
|
|
|
|
|
Because of duplicated g_signal_connect for request-aux-sender signal,
|
|
|
|
|
|
wrong stream pointer is passed to the signal handler.
|
|
|
|
|
|
Instead of passing each stream, pass stream array and get the relevant stream.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=747839
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* acinclude.m4:
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
Update autogen.sh to latest version from common
|
|
|
|
|
|
Fixes build after aclocal_check etc. helpers have been removed.
|
|
|
|
|
|
|
|
|
|
|
|
2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From bc76a8b to c8fb372
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Limit the queues to 1 buffer
|
|
|
|
|
|
We only need them to be able to pre-roll, queueing up more data here
|
|
|
|
|
|
is only going to harm latency and memory usage.
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Update comment and ASCII art to the latest code
|
|
|
|
|
|
We have a queue in front of the udpsink too to prevent the pipeline from
|
|
|
|
|
|
locking up.
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-media: Properly return first rtptime
|
|
|
|
|
|
Instead we where returning first GstBuffer timestamp. This would result
|
|
|
|
|
|
in clock skew and unwanted behaviour in RTSP playback.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=746479
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Don't leave buffer mapped
|
|
|
|
|
|
If the seq is NULL, the RTP buffer was left mapped. We should always
|
|
|
|
|
|
unmap the buffer.
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* README:
|
|
|
|
|
|
Fix typo in README
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
Fix double semicolons
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
|
|
|
|
|
|
This gives more accurate values than asking the payloader. There might be
|
|
|
|
|
|
queueing happening between the payloader and the sink.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=745704
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Don't seek for PLAY if the position will not change
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=745704
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Don't include payload type in the caps for framesize
|
|
|
|
|
|
When the sdp media attribute framesize are converted to caps
|
|
|
|
|
|
the <payload> should not be included.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
|
|
|
|
|
|
Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-sdp: add payload type to the sdp framesize attribute
|
|
|
|
|
|
The sdp framesize attribute is desribed in RFC6064. It is specified
|
|
|
|
|
|
for payloading of H263 and has the following form
|
|
|
|
|
|
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
|
|
|
|
|
|
should be added to the caps in a payloader and the <payload type> should
|
|
|
|
|
|
be added by the rtsp-server.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: test-uri: fix tainted variable
|
|
|
|
|
|
Insignificant but this keeps Coverity happy.
|
|
|
|
|
|
CID #1268404
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-netclock-client.c:
|
|
|
|
|
|
* examples/test-netclock.c:
|
|
|
|
|
|
examples: Add a simple example of network synch for live streams.
|
|
|
|
|
|
An example server and client that works for synchronising live streams
|
|
|
|
|
|
only - as it can't support pause/play.
|
|
|
|
|
|
|
|
|
|
|
|
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
rtsp-media-factory: Add functions to set/get the media gtype
|
|
|
|
|
|
Allow specifying the GType of a GstRtspMedia subclass to create
|
|
|
|
|
|
as a simpler way to get the factory to create a custom
|
|
|
|
|
|
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: fix double unlock in _get_buffer_size()
|
|
|
|
|
|
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
|
|
|
|
|
|
because of double g_mutex_unlock () usage.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=745434
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
rtsp-session: Use monotonic time for RTSP session timeout
|
|
|
|
|
|
Changed RTSP session timeout handling to monotonic time
|
|
|
|
|
|
and deprecating the API for current system time.
|
|
|
|
|
|
This fixes timeouts when the system time changes.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=743346
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-client: Only error out in PLAY if seeking actually failed
|
|
|
|
|
|
If the media was just not seekable, we continue from whatever position we are
|
|
|
|
|
|
and let the client decide if that is what is wanted or not.
|
|
|
|
|
|
Only if the actual seek failed, we can't really recover and should error out.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Add necessary queues between tee and multiudpsink
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=744379
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: If seeking fails, don't wait forever for the media to preroll again
|
|
|
|
|
|
Instead error out properly the same way as if the SEEKING query already
|
|
|
|
|
|
failed.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-stream: minor code formatting fix
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: fix logic for collect_streams
|
|
|
|
|
|
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
|
|
|
|
|
|
all streams it knows if it got any, and can check if the transport mode is OK.
|
|
|
|
|
|
CID #1268400
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Don't set the transport mode based on what elements we find
|
|
|
|
|
|
Just print a warning if the one that was set before disagrees with what
|
|
|
|
|
|
elements we found. It must already be set to something before as this
|
|
|
|
|
|
function is called after we received the SDP from ANNOUNCE in RECORD mode,
|
|
|
|
|
|
and we would reject ANNOUNCE if the RECORD flag was not set.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: rtspserver: rename shadowed variable
|
|
|
|
|
|
We have two different 'sink' variables here,
|
|
|
|
|
|
rename one of them for clarity.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: fix awkward if clause
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: test-uri: improve uri argument handling and accept file names
|
|
|
|
|
|
Print an error if the argument passed is not a URI and can't
|
|
|
|
|
|
be converted into one, or no arguments have been provided.
|
|
|
|
|
|
|
|
|
|
|
|
2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: test-uri: don't remove mount point after 10 seconds
|
|
|
|
|
|
It's very irritating when trying to test stuff repeatedly
|
|
|
|
|
|
and serves no real purpose other than showing that it can
|
|
|
|
|
|
be done.
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
examples: add new test-record to .gitignore
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtsp-media: Use flags to distinguish between PLAY and RECORD media
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
test-record: Set latency for playback-style example to 2s instead of 200ms
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: add some unit tests for ANNOUNCE and RECORD
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=743175
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: fix a couple of leaks in handle_announce
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.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:
|
|
|
|
|
|
rtsp-media: Expose latency setting for setting the rtpbin latency
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
|
|
|
|
|
|
|
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|
|
|
2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-record.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* 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-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
Add initial support for RECORD
|
|
|
|
|
|
We currently only support media that is RECORD or PLAY only, not both at once.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=743175
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: RTCP and RTP transport cache cookies seperated
|
|
|
|
|
|
RTCP packets were not sent because the same tr_cache_cookie was used for
|
|
|
|
|
|
both RTP and RTCP. So only one of the tr_cache lists were populated
|
|
|
|
|
|
depending on which one was sent first. If the tr_cache list is not
|
|
|
|
|
|
populated then no packets can be sent. Most often this happened to be
|
|
|
|
|
|
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
|
|
|
|
|
|
resulted in both the tr_cache_lists to be populated regardless of which
|
|
|
|
|
|
one was sent first.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: fix false compiler warning
|
|
|
|
|
|
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: log interleaved data received
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Use a random session ID in the SDP
|
|
|
|
|
|
RFC4566 Section 5.2 says that it should make the username, session id,
|
|
|
|
|
|
nettype, addrtype and unicast address tuple globally unique. Always using
|
|
|
|
|
|
1188340656180883 is not going to guarantee that: https://xkcd.com/221/
|
|
|
|
|
|
Instead let's create a 64 bit random number, which at least brings us
|
|
|
|
|
|
closer to the goal of global uniqueness.
|
|
|
|
|
|
https://tools.ietf.org/html/rfc4566#section-5.2
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: Don't call gst_init() and gst_get_option_group()
|
|
|
|
|
|
The latter calls the former at the appropriate time.
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Drop trailing \0 of RTSP DATA messages
|
|
|
|
|
|
We add a trailing \0 in GstRTSPConnection to make parsing of
|
|
|
|
|
|
string message bodies easier (e.g. the SDP from DESCRIBE) but
|
|
|
|
|
|
for actual data this means we have to drop it or otherwise
|
|
|
|
|
|
create invalid data.
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
|
|
|
|
|
|
Fixes crash when two threads access handle_new_sample() at the same
|
|
|
|
|
|
time, one for RTP, one for RTCP.
|
|
|
|
|
|
Otherwise, when iterating over the transports cache, it might be modified by
|
|
|
|
|
|
another thread at the same time if the transports cookie has changed.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=742954
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Set format=TIME on our app sources for TCP
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
|
|
|
|
|
|
This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
|
|
|
|
|
|
RFC 2326 states that session IDs may consist of alphanumeric as well as
|
|
|
|
|
|
the safe characters $-_.+ -- N.B. the percent character is not allowed.
|
|
|
|
|
|
Previously the session ID was URI-escaped, this meant that any character
|
|
|
|
|
|
which was not alphanumeric or any of the characters +-._~ would be
|
|
|
|
|
|
percent encoded. While the RFC (surprisingly) mentions that linear white
|
|
|
|
|
|
space in session IDs should be URI-escaped, it does not say anything
|
|
|
|
|
|
about other characters. Moreover no white space is allowed in the
|
|
|
|
|
|
session ID. Finally the percent character which is the result of
|
|
|
|
|
|
URI-escaping is not allowed in a session ID.
|
|
|
|
|
|
So there is no reason to do any URI-escaping, and now it is removed.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=742869
|
|
|
|
|
|
|
|
|
|
|
|
2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f2c6b95 to bc76a8b
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
Fix 'make check' from top-level directory
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: Add command-line parsing and take a 'port' argument
|
|
|
|
|
|
This allows users to run multiple servers on different ports for testing.
|
|
|
|
|
|
Only done for examples that actually take arguments and hence are capable of
|
|
|
|
|
|
outputting different streams for each instance on each port.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=742115
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
rtsp-client: Add a send_message default signal handler
|
|
|
|
|
|
This allows subclasses to easily hook into the response sending
|
|
|
|
|
|
mechanism without doing everything from a signal, which seems
|
|
|
|
|
|
awkward from subclasses.
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From ef1ffdc to f2c6b95
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: add --disable-examples switch
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=741678
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-video-rtx.c:
|
|
|
|
|
|
examples: add a retransmisison example implementing RFC4588
|
|
|
|
|
|
Currently only SSRC-multiplexed rtx streams are supported
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Fix some minor memory leaks
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Some minor cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Fix compiler warnings
|
|
|
|
|
|
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
|
|
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
|
|
^
|
|
|
|
|
|
rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
|
|
|
|
|
|
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
|
|
|
|
|
^
|
|
|
|
|
|
|
|
|
|
|
|
2014-11-27 01:12:36 +1100 Matthew Waters <matthew@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-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
media: implement ssrc-multiplexed retransmission support
|
|
|
|
|
|
based off RFC 4588 and the server-rtpaux example in -good
|
|
|
|
|
|
|
|
|
|
|
|
2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp: Ref transports in hash table.
|
|
|
|
|
|
Also ref streams for transports.
|
|
|
|
|
|
This solves a crash when reciving a rtcp after teardown but before
|
|
|
|
|
|
client finalize.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
|
|
|
|
|
|
|
|
|
|
|
|
2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 7bb2bce to ef1ffdc
|
|
|
|
|
|
|
|
|
|
|
|
2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: refactor cleanup of cached media
|
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|
2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
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|
|
* tests/check/gst/client.c:
|
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|
|
tests: Remove FIXME
|
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|
|
The session leak is now fixed, lets remove those FIXME comments.
|
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|
2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
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* tests/check/gst/rtspserver.c:
|
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|
|
tests: Test to setup two sessions on one connection
|
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|
|
https://bugzilla.gnome.org/show_bug.cgi?id=739112
|
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|
2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
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|
|
* tests/check/gst/rtspserver.c:
|
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|
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|
tests: Test setup with tcp transport
|
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|
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|
https://bugzilla.gnome.org/show_bug.cgi?id=739112
|
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2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
|
|
client: Configure transport after creating session media
|
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|
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|
|
The default implementation of configure_client_transport() in
|
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|
|
rtsp-client uses the session media when it chooses channels for
|
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|
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|
|
interleaved traffic.
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|
https://bugzilla.gnome.org/show_bug.cgi?id=739112
|
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2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
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|
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-session-media.c:
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client: Stop caching media in client when doing setup
|
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|
If the media has been managed by a session media, it should not be
|
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|
|
cached in the client any longer. The GstRTSPSessionMedia object is now
|
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|
|
responsible for unpreparing the GstRTSPMedia object using
|
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|
|
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
|
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|
session media.
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|
https://bugzilla.gnome.org/show_bug.cgi?id=739112
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|
2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
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|
* gst/rtsp-server/rtsp-stream.c:
|
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|
|
rtsp-stream: unref srtp decoder when leaving bin
|
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|
|
https://bugzilla.gnome.org/show_bug.cgi?id=739481
|
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|
2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
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* gst/rtsp-server/rtsp-client.c:
|
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|
|
|
|
rtsp-client: mikey memory leaks
|
|
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|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=739383
|
|
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|
2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
|
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|
|
|
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|
|
* common:
|
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|
|
|
|
Automatic update of common submodule
|
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|
|
From 84d06cd to 7bb2bce
|
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|
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|
|
2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
|
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|
|
|
|
|
* Makefile.am:
|
|
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|
|
|
Parallelise 'make check-valgrind'
|
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|
2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
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|
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|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From a8c8939 to 84d06cd
|
|
|
|
|
|
|
|
|
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|
|
2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 36388a1 to a8c8939
|
|
|
|
|
|
|
|
|
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|
|
2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: deactivate media when shutting down from paused
|
|
|
|
|
|
This was only done when going directly from playing.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
|
|
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|
|
2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-context.h:
|
|
|
|
|
|
rtsp-client: add stream transport to context
|
|
|
|
|
|
We add the stream transport to the context so we can get the configured
|
|
|
|
|
|
client stream transport in the setup request signal.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
|
|
|
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|
|
|
|
2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: release lock even not all transports have been removed
|
|
|
|
|
|
We don't want to keep the lock even we return FALSE because not all the
|
|
|
|
|
|
transports have been removed. This could lead into a deadlock.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=737797
|
|
|
|
|
|
|
|
|
|
|
|
2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
|
|
|
|
|
|
These were renamed in GstRTPBasePayload in 1.0
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: set session media to NULL without the lock
|
|
|
|
|
|
We need to set session medias to NULL without the client lock otherwise
|
|
|
|
|
|
we can end up in a deadlock if another thread is waiting for the lock
|
|
|
|
|
|
and media unprepare is also waiting for that thread to end.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=737690
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Set state to UNPREPARING in all cases
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: set state to unpreparing when unprepare is initiated
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=737675
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Remove backlog limit while processings requests
|
|
|
|
|
|
If the backlog limit is kept two cases of deadlocks may be
|
|
|
|
|
|
encountered when streaming over TCP. Without the backlog
|
|
|
|
|
|
limit this deadlocks can not happen, at the expence of
|
|
|
|
|
|
memory usage.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: do not free main context before rtsp watch
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=737110
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: Extend unit test timeout to accomodate for valgrind
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
rtsp-*: Treat sending packets to clients as keepalive
|
|
|
|
|
|
As long as gst-rtsp-server can successfully send RTP/RTCP data to
|
|
|
|
|
|
clients then the client must be reading. This change makes the server
|
|
|
|
|
|
timeout the connection if the client stops reading.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Allow backlog to grow while expiring session
|
|
|
|
|
|
Allow the send backlog in the RTSP watch to grow to unlimited size while
|
|
|
|
|
|
attempting to bring the media pipeline to NULL due to a session
|
|
|
|
|
|
expiring. Without this change the appsink element cannot change state
|
|
|
|
|
|
because it is blocked while rendering data in the new_sample callback.
|
|
|
|
|
|
This callback will block until it has successfully put the data into the
|
|
|
|
|
|
send backlog. There is a chance that the send backlog is full at this
|
|
|
|
|
|
point which means that the callback may block for a long time, possibly
|
|
|
|
|
|
forever. Therefore the media pipeline may also be prevented from
|
|
|
|
|
|
changing state for a long time.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Make old compilers happy
|
|
|
|
|
|
rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
|
|
|
|
|
|
Just in case that guint8 doesn't fit in a pointer. Just in case ...
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: raise the backlog limits before pausing
|
|
|
|
|
|
We need to raise the backlog limits before pausing the pipeline or else
|
|
|
|
|
|
the appsink might be blocking in the render method in wait_backlog() and
|
|
|
|
|
|
we would deadlock waiting for paused.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: make define for the WATCH_BACKLOG
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=736322
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: simplify session transport handling
|
|
|
|
|
|
link/unlink of the transport in a session was done to keep track of all
|
|
|
|
|
|
TCP transports and to send RTP/RTCP data to the streams. We can simplify
|
|
|
|
|
|
that by putting all the TCP transports in a hashtable indexed with the
|
|
|
|
|
|
channel number.
|
|
|
|
|
|
We also don't need to link/unlink the transports when we pause/resume
|
|
|
|
|
|
the streams. The same effect is already achieved when we pause/play the
|
|
|
|
|
|
media. Indeed, when we pause the media, the transport is removed from
|
|
|
|
|
|
the media and the callbacks will not be called anymore.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=736041
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
stream-transport: make method to handle received data
|
|
|
|
|
|
Make a method to handle the data received on a channel. It sends the
|
|
|
|
|
|
data to the stream of the transport on the RTP or RTCP pads based on
|
|
|
|
|
|
the channel number.
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
test: add example of dumping RTCP reports
|
|
|
|
|
|
|
|
|
|
|
|
2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-media: Make sure that sequence numbers are monotonic after pause
|
|
|
|
|
|
The sequence number is not monotonic for RTP packets after pause. The
|
|
|
|
|
|
reason is basepayloader generates a randon sequence number when the
|
|
|
|
|
|
pipeline goes from ready to pause. With this fix generation of sequence
|
|
|
|
|
|
number will be monotonic when going from pause to play request.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=736017
|
|
|
|
|
|
|
|
|
|
|
|
2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Protect saved clients watch with a mutex
|
|
|
|
|
|
Fixes a crash when close() is called while merging clients
|
|
|
|
|
|
in handle_tunnel(). In that case close() would destroy the
|
|
|
|
|
|
watch while it is still being used in handle_tunnel().
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=735570
|
|
|
|
|
|
|
|
|
|
|
|
2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Remove the multicast group udp sources when removing from the bin
|
|
|
|
|
|
|
|
|
|
|
|
2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp-media: Query position and stop time only on the RTP parts of the pipeline
|
|
|
|
|
|
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
|
|
|
|
|
|
seeking and will always continue counting the time. This leads to
|
|
|
|
|
|
the NPT after a backwards seek to be something completely different
|
|
|
|
|
|
to the actual seek position.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=732644
|
|
|
|
|
|
|
|
|
|
|
|
2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-appsrc.c:
|
|
|
|
|
|
examples: fix another reference leak
|
|
|
|
|
|
gst_rtsp_media_get_element() returns a new ref.
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-appsrc.c:
|
|
|
|
|
|
examples: unref element after usage
|
|
|
|
|
|
gst_bin_get_by_name_recurse_up() returns an element
|
|
|
|
|
|
reference that must be unreffed after usage.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=734546
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
signals: Fix copy-pasto in target-state signal offset
|
|
|
|
|
|
|
|
|
|
|
|
2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Makefile: Add usage of build-checks step
|
|
|
|
|
|
Allows building checks without running them
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Listen on the multicast group for RTP/RTCP packets
|
|
|
|
|
|
When a UDP multicast transport is used it is expected that the server listens
|
|
|
|
|
|
for RTP and RTCP packets on the multicast group with the corresponding port.
|
|
|
|
|
|
Without this we will never get RTCP packets from clients in multicast mode.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=732238
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2014-07-19 15:56:31 +00:00
|
|
|
|
=== release 1.4.0 ===
|
|
|
|
|
|
|
2015-06-07 09:20:01 +00:00
|
|
|
|
2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-07-19 15:56:31 +00:00
|
|
|
|
|
2015-06-07 09:20:01 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2014-07-19 15:56:31 +00:00
|
|
|
|
* configure.ac:
|
2015-06-07 09:20:01 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.4.0
|
2014-07-19 15:56:31 +00:00
|
|
|
|
|
|
|
|
|
|
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: correct misspelled words in description
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=733244
|
|
|
|
|
|
|
2014-07-11 10:19:08 +00:00
|
|
|
|
=== release 1.3.91 ===
|
|
|
|
|
|
|
2014-07-19 15:56:31 +00:00
|
|
|
|
2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-07-11 10:19:08 +00:00
|
|
|
|
|
2014-07-19 15:56:31 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2014-07-11 10:19:08 +00:00
|
|
|
|
* configure.ac:
|
2014-07-19 15:56:31 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.3.91
|
2014-07-11 10:19:08 +00:00
|
|
|
|
|
|
|
|
|
|
2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
docs: update docs
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: implement client REMOVE filter
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: expose _close() method
|
|
|
|
|
|
Expose a previously internal close method to close the client
|
|
|
|
|
|
connection.
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
session-pool: signal session-removed outside of the lock
|
|
|
|
|
|
Release the lock before emiting the session-removed signal.
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
filter: Release lock in filter functions
|
|
|
|
|
|
Release the object lock before calling the filter functions. We need to
|
|
|
|
|
|
keep a cookie to detect when the list changed during the filter
|
|
|
|
|
|
callback. We also keep a hashtable to make sure we only call the filter
|
|
|
|
|
|
function once for each object in case of concurrent modification.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: check if watch is set in handle_teardown()
|
|
|
|
|
|
The unit tests run without a watch
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
client tests: send teardown to cleanup session
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
server tests: send teardown to cleanup session
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: keep ref to client for the session removed handler
|
|
|
|
|
|
This extra ref will be dropped when all client sessions have been
|
|
|
|
|
|
removed. A session is removed when a client sends teardown, closes its
|
|
|
|
|
|
endpoint of the TCP connection or the sessions expires.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
client: manage media in session as a last step
|
|
|
|
|
|
Once we manage a media in a session, we can't unmanage it anymore
|
|
|
|
|
|
without destroying it. Therefore, first check everything before we
|
|
|
|
|
|
manage the media, otherwise if something is wrong we have no way to
|
|
|
|
|
|
unmanage the media.
|
|
|
|
|
|
If we created a new session and something went wrong, remove the session
|
|
|
|
|
|
again. Fixes a leak in the unit test.
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
examples: print 'stream ready at url' for mp4 and ogg example
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp: fix for MIKEY api change
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: free watch context only once
|
|
|
|
|
|
The watch context is freed when the source is destroyed. Avoids
|
|
|
|
|
|
a CRITICAL when we try to unref the context twice.
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix build
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: protect sessions with lock
|
|
|
|
|
|
Protect the list of sessions with the lock.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
|
|
|
|
|
|
|
|
|
|
|
|
2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Client: keep a ref to the session
|
|
|
|
|
|
Don't just keep a weak ref to the session objects but use a hard ref. We
|
|
|
|
|
|
will be notified when a session is removed from the pool (expired) with
|
|
|
|
|
|
the new session-removed signal.
|
|
|
|
|
|
Don't automatically close the RTSP connection when all the sessions of
|
|
|
|
|
|
a client are removed, a client can continue to operate and it can create
|
|
|
|
|
|
a new session if it wants. If you want to remove the client from the
|
|
|
|
|
|
server, you have to use gst_rtsp_server_client_filter() now.
|
|
|
|
|
|
Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=732226
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
session-pool: add session-removed signal
|
|
|
|
|
|
Add a signal to be notified when a session is removed from the pool.
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Make rtsp-server.h a single-include header, use it for G-I
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=732411
|
|
|
|
|
|
|
2014-06-28 09:48:29 +00:00
|
|
|
|
=== release 1.3.90 ===
|
|
|
|
|
|
|
2014-07-11 10:19:08 +00:00
|
|
|
|
2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-06-28 09:48:29 +00:00
|
|
|
|
|
2014-07-11 10:19:08 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2014-06-28 09:48:29 +00:00
|
|
|
|
* configure.ac:
|
2014-07-11 10:19:08 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.3.90
|
2014-06-28 09:48:29 +00:00
|
|
|
|
|
|
|
|
|
|
2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: crypto can be NULL
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
introspection: add missing allow-none annotations
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=730952
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
introspection: add (nullable) annotations to return values
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=730952
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
gi: improve annotations
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
signals: use generic marshal function
|
|
|
|
|
|
Use the generic C marshal function.
|
|
|
|
|
|
Use more explicit type instead of G_TYPE_POINTER
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-context.h:
|
|
|
|
|
|
context: add type macro
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
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|
|
|
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|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
sdp: hide key length defines
|
|
|
|
|
|
They don't have a namespace.
|
|
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|
2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2014-06-22 17:36:14 +00:00
|
|
|
|
=== release 1.3.3 ===
|
|
|
|
|
|
|
2014-06-28 09:48:29 +00:00
|
|
|
|
2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-06-22 17:36:14 +00:00
|
|
|
|
|
2014-06-28 09:48:29 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2014-06-22 17:36:14 +00:00
|
|
|
|
* configure.ac:
|
2014-06-28 09:48:29 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.3.3
|
2014-06-22 17:36:14 +00:00
|
|
|
|
|
|
|
|
|
|
2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
mikey: add different key length parameters
|
|
|
|
|
|
Add encryption and authentication key length parameters to MIKEY. For
|
|
|
|
|
|
the encoders, the key lengths are obtained from the cipher and auth
|
|
|
|
|
|
algorithms set in the caps. For the decoders, they are obtained while
|
|
|
|
|
|
parsing the key management from the client.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/stream.c:
|
|
|
|
|
|
stream tests: Make sure we get right multicast address from stream
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
|
|
|
|
|
|
|
|
|
|
|
|
2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: ref the context until rtsp watch is alive
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
|
|
|
|
|
|
|
|
|
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|
|
2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Destroy the rtsp watch after connection close
|
|
|
|
|
|
|
|
|
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|
|
2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix confusing comment
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
rtsp-session: Timeout in header.
|
|
|
|
|
|
Adding the possbilty to always have timout in header.
|
|
|
|
|
|
This is configurabe with setting "timeout-always-visible".
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
|
|
|
|
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|
|
|
|
|
|
|
2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2014-05-21 11:06:36 +00:00
|
|
|
|
=== release 1.3.2 ===
|
|
|
|
|
|
|
2014-06-22 17:36:14 +00:00
|
|
|
|
2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-05-21 11:06:36 +00:00
|
|
|
|
|
2014-06-22 17:36:14 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
|
|
|
|
|
* common:
|
2014-05-21 11:06:36 +00:00
|
|
|
|
* configure.ac:
|
2014-06-22 17:36:14 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.3.2
|
2014-05-21 11:06:36 +00:00
|
|
|
|
|
|
|
|
|
|
2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 211fa5f to 1f5d3c3
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: store TCP ports in transport
|
|
|
|
|
|
Store the TCP ports in the transport when we are doing RTSP over TCP.
|
|
|
|
|
|
This way, we can easily get to the ports from the transport.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: add signals for new RTP/RTCP encoders
|
|
|
|
|
|
New signals to allow the user to configure the dynamically created
|
|
|
|
|
|
encoders.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=730228
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: Make suspend()/unsuspend() virtual
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix send-message signal marshaller
|
|
|
|
|
|
Use generic marshalling for the send-message signal. It has
|
|
|
|
|
|
two POINTER arguments, not just one.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=729900
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
tests: add and remove pads only once
|
|
|
|
|
|
In this test we simulate a dynamic pad by watching the caps event.
|
|
|
|
|
|
Because of renegotiation in the base payloader now, this caps is sent
|
|
|
|
|
|
multiple times but we can only deal with 1 invocation, use a variable to
|
|
|
|
|
|
only 'add and remove' the pad once.
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: add unit test for correct handling of Require headers
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=729426
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
|
|
|
|
|
|
Servers must handle Require headers and must report a failure
|
|
|
|
|
|
if they don't handle any of the Required options, see RFC 2326,
|
|
|
|
|
|
section 12.32: https://tools.ietf.org/html/rfc2326#page-54
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=729426
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
|
2014-05-03 16:40:24 +00:00
|
|
|
|
=== release 1.3.1 ===
|
|
|
|
|
|
|
2014-05-21 11:06:36 +00:00
|
|
|
|
2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
|
2014-05-03 16:40:24 +00:00
|
|
|
|
|
2014-05-21 11:06:36 +00:00
|
|
|
|
* ChangeLog:
|
|
|
|
|
|
* NEWS:
|
|
|
|
|
|
* RELEASE:
|
2014-05-03 16:40:24 +00:00
|
|
|
|
* configure.ac:
|
2014-05-21 11:06:36 +00:00
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
Release 1.3.1
|
2014-05-03 16:40:24 +00:00
|
|
|
|
|
|
|
|
|
|
2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From bcb1518 to 211fa5f
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
Update .gitignore
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/sessionmedia.c:
|
|
|
|
|
|
tests: fix memory leak in sessionmedia unit test
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: emit a signal before sending a message
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: pass context to send_message
|
|
|
|
|
|
Pass the current context to send_message, we will need it later.
|
|
|
|
|
|
|
|
|
|
|
|
2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix typo in comment
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: Do not stop thread twice if default_prepare() fails
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: set the watch to flushing before going to NULL
|
|
|
|
|
|
First set the watch to flushing so that we unblock any current and
|
|
|
|
|
|
future attempt to send data on the watch, Then set the pipeline to
|
|
|
|
|
|
NULL.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* tests/check/gst/sessionpool.c:
|
|
|
|
|
|
rtsp-session-pool: Fixes annotation
|
|
|
|
|
|
Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
|
|
|
|
|
|
in the sessionpool test.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: make media_prepare virtual
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media: stop the thread in more error cases
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media: allow NULL as the thread
|
|
|
|
|
|
Use the default context whan passing a NULL thread.
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: indent cleanup
|
|
|
|
|
|
Coverity was moaning about unreachable code, and I think it was just
|
|
|
|
|
|
confused by { being before the label. We'll see if it pops up again.
|
|
|
|
|
|
Coverity 1197705
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
client: Add drop-backlog property
|
|
|
|
|
|
When we have too many messages queued for a client (currently hardcoded
|
|
|
|
|
|
to 100) we overflow and drop the messages. Add a drop-backlog property
|
|
|
|
|
|
to control this behaviour. Setting this property to FALSE will retry
|
|
|
|
|
|
to send the messages to the client by waiting for more room in the
|
|
|
|
|
|
backlog.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: support for POST before GET when setting up a tunnel
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: remove watch of the second client after http tunnel setup
|
|
|
|
|
|
The second client will be freed after the HTTP tunnel has been set up.
|
|
|
|
|
|
Make sure it's RTSP watch is never dispatched again.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media: Make media_prepare() fail if port allocation fails
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media test: cleanup the thread pool in tests
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
rtsp-media: Unblock blocked streams in unprepare
|
|
|
|
|
|
The streams will be blocked when a live media is prepared.
|
|
|
|
|
|
The streams should be unblocked in gst_rtsp_media_unprepare.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: release the state lock when going to NULL
|
|
|
|
|
|
Set our state to UNPREPARING and release the state-lock before
|
|
|
|
|
|
setting the pipeline to the NULL state. This way, any pad-added
|
|
|
|
|
|
callback will be able to take the state-lock and check that we are now
|
|
|
|
|
|
unpreparing instead of deadlocking.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: protect status with lock
|
|
|
|
|
|
Make sure we only update the status with the lock.
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp: update for MIKEY API changes
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: parse the mikey response from the client
|
|
|
|
|
|
Parse the mikey response from the client and update the policy for
|
|
|
|
|
|
each SSRC.
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add method to set crypto info
|
|
|
|
|
|
Make a method to configure the crypto information of a stream.
|
|
|
|
|
|
Set udpsrc in READY instead of PAUSED so that we can configure caps
|
|
|
|
|
|
later.
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: cleanup error paths
|
|
|
|
|
|
|
|
|
|
|
|
2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix docs
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
test: enable SRTP only on RTSPS
|
|
|
|
|
|
We only want to enable SRTP when doing rtsp over TLS so that we can
|
|
|
|
|
|
exchange the keys in a secure way.
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
test: print an error on failure
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
stream: add SRTP support
|
|
|
|
|
|
Install srtp encoder and decoder elements in rtpbin
|
|
|
|
|
|
Add MIKEY in SDP
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/sessionpool.c:
|
|
|
|
|
|
tests: Add unit tests for sessionpool
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/threadpool.c:
|
|
|
|
|
|
tests: Improve code coverage of rtsp-threadpool tests
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/sessionmedia.c:
|
|
|
|
|
|
tests: Improve code coverage for rtsp-session-media
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
gobject-introspection: Add annotations to support language bindings
|
|
|
|
|
|
In addition a few cosmetic changes:
|
|
|
|
|
|
* Adjust the order of arguments
|
|
|
|
|
|
* Fix typo: occured -> occurred
|
|
|
|
|
|
* Fix indentation after Return:-clauses
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Don't mix IPv4 and IPv6 addresses
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: take caps after the session manager
|
|
|
|
|
|
Take the caps for the SDP after they leave the rtpbin so that we can
|
|
|
|
|
|
also get the properties added by rtpbin elements.
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: release lock while pushing out packets
|
|
|
|
|
|
Keep a cache of the transports and use this to iterate the transport
|
|
|
|
|
|
while pushing packets. This allows us to release the lock early.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=725898
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
rtsp-client: vmethod for modifying tunnel GET response
|
|
|
|
|
|
Add a vmethod tunnel_http_response where the response to the HTTP GET
|
|
|
|
|
|
for tunneled connections can be modified.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: make 1 media line per profile
|
|
|
|
|
|
If we have multiple profiles (AVP or AVPF) for a stream, make one m=
|
|
|
|
|
|
line in the SDP for each profile. The client is then supposed to pick
|
|
|
|
|
|
one of the profiles in the SETUP request. Because the m= lines have the
|
|
|
|
|
|
same pt, the client also knows that only 1 option is possible.
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
factory: add profile property and pass to media and streams
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-multicast.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: pass multicast connection for multicast-only stream
|
|
|
|
|
|
Pass the multicast address of the stream in the connection info in the
|
|
|
|
|
|
SDP so that clients try a multicast connection first.
|
|
|
|
|
|
Only allow multicast connections in the test-multicast example. Also
|
|
|
|
|
|
increase the TTL a little.
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
.gitignore: Ignore gcov intermediate files
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: release some locks in error cases
|
|
|
|
|
|
|
|
|
|
|
|
2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
docs: Enable and fix gtk-doc warnings
|
|
|
|
|
|
* Makefile: Enable gtk-doc warnings, like the rest of GStreamer
|
|
|
|
|
|
* addresspool/mediafactory: Add missing annotation colon
|
|
|
|
|
|
* stream: Annotate return value
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From fe1672e to bcb1518
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 1a07da9 to fe1672e
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
examples: use LDADD for libs instead of LDFLAGS
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: make sure releases are in .doap file
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
examples: test-cgroups: don't put code with side effects into g_assert()
|
|
|
|
|
|
The g_assert() might get compiled out with the right
|
|
|
|
|
|
compiler/preprocessor flags.
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
examples: add cgroup test binary to .gitignore
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
examples: fix cgroup test build
|
|
|
|
|
|
Fixes build failure caused by compiler warning:
|
|
|
|
|
|
test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
.gitignore: ignore temp files created in the course of 'make check'
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: don't loose frames handling new PLAY request
|
|
|
|
|
|
If client supplied a range check if the range specifies the start point.
|
|
|
|
|
|
If not, then do an accurate seek to the current position. If a start
|
|
|
|
|
|
point was specified do do a key unit seek to make sure the streaming
|
|
|
|
|
|
starts with decodeable frames.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Revert "media: only flush when setting a new start position"
|
|
|
|
|
|
This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
|
|
|
|
|
|
We need to do the flush in all cases, demuxer block currently for
|
|
|
|
|
|
non-flushing seeks.
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: only flush when setting a new start position
|
|
|
|
|
|
Only flush the pipeline when we change the start position with
|
|
|
|
|
|
a seek.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=724611
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: set ttl-mc before adding the socket
|
|
|
|
|
|
Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
|
|
|
|
|
|
never be set on socket.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: stop thread if media is already prepared
|
|
|
|
|
|
in gst_rtsp_media_prepare() the thread is not used if media is already
|
|
|
|
|
|
prepared (e.g. media shared) so we want to stop the thread. otherwise, a
|
|
|
|
|
|
leak occurs.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=724182
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
build: Ship gst-rtsp-server.doap file
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: Fix another compiler warning with gcc
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
rtsp-server: Fix lots of compiler warnings with clang
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst-rtsp-server.doap:
|
|
|
|
|
|
* tests/Makefile.am:
|
|
|
|
|
|
configure: Synchronise with the configure scripts of the other modules
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
Revert "rtsp-server: support build against last stable release"
|
|
|
|
|
|
This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
|
|
|
|
|
|
Let us require 1.2.3 now, which is going to be released in a few
|
|
|
|
|
|
minutes.
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
session: improve RTP-Info
|
|
|
|
|
|
Ignore streams that can't generate RTP-Info instead of failing.
|
|
|
|
|
|
Don't return the empty string when all streams are unconfigured but
|
|
|
|
|
|
return NULL so that we don't generate and empty RTP-Info header.
|
|
|
|
|
|
Improve docs a little.
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
Don't free rtpinfo GString when it is NULL
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: only set keyframe flag when modifying start
|
|
|
|
|
|
Only set the keyframe flag when we modify the start position. The
|
|
|
|
|
|
keyframe flag should probably be ignored when no change is requested but
|
|
|
|
|
|
until we can claim this is all documented properly and all demuxer
|
|
|
|
|
|
implement this, avoid setting the flag.
|
|
|
|
|
|
See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
thread-pool: Unref source after mainloop has quit to avoid races in GLib
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: handle NULL seqnum and rtptime arguments
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* tests/check/gst/threadpool.c:
|
|
|
|
|
|
thread-pool: Unref reused threads in gst_rtsp_thread_stop()
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
|
|
|
|
|
|
|
|
|
|
|
|
2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: add fallback for missing stats property
|
|
|
|
|
|
Use a fallback when the payloader does not have a stats property
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f7bc1c3 to 1a07da9
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: don't leak stats structure
|
|
|
|
|
|
Don't leak the stats structure and deal with NULL stats.
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: Get rtpinfo properties atomically from payloader
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: refactor state change functions and signals
|
|
|
|
|
|
Make functions to set the target state and the pipeline state and emit
|
|
|
|
|
|
the signals from those functions.
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add signal to notify of pending state changes
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-server: support build against last stable release
|
|
|
|
|
|
Until 1.2.3 is out with the new get_type function and we
|
|
|
|
|
|
can require that.
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: fix compilation
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add property to configure profiles
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: let stream check supported transport
|
|
|
|
|
|
Delegate the check if a transport is allowed to the stream.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=720696
|
|
|
|
|
|
|
|
|
|
|
|
2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add method to check supported transport
|
|
|
|
|
|
Add a method to check if a transport is supported
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure.ac: Only check for gstreamer-check, not check
|
|
|
|
|
|
We include check in gstreamer-check since quite some time now.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: return clock-rate from get_rtpinfo
|
|
|
|
|
|
And use it to correct the rtptime to the requested start-time.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=712198
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
session-media: calculate start-time
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: also return the running-time
|
|
|
|
|
|
Return the running-time in the rtpinfo as well.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
session-media: let the session-media make the RTPInfo
|
|
|
|
|
|
Add method to create the RTPInfo for a stream-transport.
|
|
|
|
|
|
Add method to create the RTPInfo for all stream-transports in a
|
|
|
|
|
|
session-media.
|
|
|
|
|
|
Use the session-media RTPInfo code in client. This allows us to refactor
|
|
|
|
|
|
another method to link the TCP callbacks.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
mount-points: sort sequence before g_sequence_lookup
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
|
|
|
|
|
|
sort sequence if dirty, otherwise lookup will fail.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: rename package from gst-rtsp to gst-rtsp-server
|
|
|
|
|
|
To match git module name and avoid confusion with the
|
|
|
|
|
|
rtsp lib in gst-plugins-base and rtsp plugin in -good.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump core/base/good requirement to 1.2.0
|
|
|
|
|
|
Bump to released stable version and make implicit
|
|
|
|
|
|
requirements explicit.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Fix broken gettext setup which is not used anyway
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From dbedaa0 to d48bed3
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add setup_sdp vmethod
|
|
|
|
|
|
gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
|
|
|
|
|
|
gst_rtsp_media_setup_sdp.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: Check return value of sscanf
|
|
|
|
|
|
streamid is only valid if sscanf matched something.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Fix iteration
|
|
|
|
|
|
Wouldn't even enter the code block otherwise (i++ was used as the check
|
|
|
|
|
|
and not the postfix).
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add vmethod to configure media and streams
|
|
|
|
|
|
Implement a vmethod that can be used to configure the media and the
|
|
|
|
|
|
streams based on the current context. Handle the blocksize handling in
|
|
|
|
|
|
the default handler.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=720667
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
Make git ignore more unit test binaries
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-context.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.h:
|
|
|
|
|
|
rtsp-server: add padding to many public structures
|
|
|
|
|
|
Not mini objects though, since they are not subclassable
|
|
|
|
|
|
anyway, nor kept on the stack or inlined in a structure.
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
media: add new create_rtpbin vmethod
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=719734
|
|
|
|
|
|
|
|
|
|
|
|
2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
tests: fix memory leak, free test's thread pool
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
stream-transport: free url in finalize
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: also do state change in suspended state
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: also handle prepare and range in suspended state
|
|
|
|
|
|
When we are suspended, we are already prepared.
|
|
|
|
|
|
We can get the range in the suspended state.
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/sessionmedia.c:
|
|
|
|
|
|
check: add test for uri in setup
|
|
|
|
|
|
Added unit tests for the new functionality in GstRTSPStreamTransport.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: store setup uri and use in PLAY response
|
|
|
|
|
|
Store the uri used when doing the setup and use that in the PLAY
|
|
|
|
|
|
response.
|
|
|
|
|
|
fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
stream-transport: add method to get/set url
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: suspend after SDP and unsuspend before PLAYING
|
|
|
|
|
|
Based on patches by Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.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-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
media: add suspend modes
|
|
|
|
|
|
Add support for different suspend modes. The stream is suspended right after
|
|
|
|
|
|
producing the SDP and after PAUSE. Different suspend modes are available that
|
|
|
|
|
|
affect the state of the pipeline. NONE leaves the pipeline state unchanged and
|
|
|
|
|
|
is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
|
|
|
|
|
|
state and RESET will bring the pipeline to the NULL state.
|
|
|
|
|
|
A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
|
|
|
|
|
|
this means that the pipeline needs to be prerolled again.
|
|
|
|
|
|
Base on patches by Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: start live streams in blocked state
|
|
|
|
|
|
Start live streams in the blocked state and make them preroll using the
|
|
|
|
|
|
messages. This ensure that no data is played by the sink until we explicitly
|
|
|
|
|
|
unblock the stream right before going to PLAYING.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: refactor starting and waiting for preroll
|
|
|
|
|
|
Based on patches from Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add API to block streams
|
|
|
|
|
|
Add an API to block on the streams and make it post a message.
|
|
|
|
|
|
Based on patch by Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
docs: Specify the override file
|
|
|
|
|
|
Even if it's empty (for now) it avoids make distcheck complaining
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: move default implementations to where they are used
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: take the right lock in gst_rtsp_media_set_pipeline_state()
|
|
|
|
|
|
We need to take the state_lock when calling this method.
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: handle add-added on non-bins too
|
|
|
|
|
|
Handle dynamic payloaders that are not bins, as used in the unit-test.
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media/-factory: Fix request pad name comments
|
|
|
|
|
|
These must be escaped for gtk-doc to parse the comments without warnings.
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
rtsp-media: remove transports if media is in error status
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
|
|
|
|
|
|
trying to change to GST_STATE_NULL and media is in error status, we
|
|
|
|
|
|
remove all transports.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: use element metadata to find payloader
|
|
|
|
|
|
Use the element metadata to find the payloader instead of checking
|
|
|
|
|
|
for the base class.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
rtsp-stream: add getter for payload type
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
|
|
|
|
|
|
element and create the stream with this one instead of the dynpay%d
|
|
|
|
|
|
element.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=712396
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-context.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
rtsp-*: Refer to NULL as a constant in comments
|
|
|
|
|
|
Plus one typo fix.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=714988
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
rtsp-*: Fix type name typos in comments
|
|
|
|
|
|
* rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
|
|
|
|
|
|
* rtsp-auth: Refer to part of constant name as text
|
|
|
|
|
|
* rtsp-auth/-permissions/-token: Refer to Permissions not Permission
|
|
|
|
|
|
* rtsp-session-media: Fix GstRTSPSessionMedia typo
|
|
|
|
|
|
* rtsp-stream: Fix typo when refering to GstBin
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=714988
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
docs: Improve documentation
|
|
|
|
|
|
* Include annotation-glossary to quiet gtk-doc
|
|
|
|
|
|
* Rename remaining ClientState -> Context
|
|
|
|
|
|
* Rename object hierarchy file
|
|
|
|
|
|
* Remove stale chapter references
|
|
|
|
|
|
* Add missing function and object references
|
|
|
|
|
|
* Include missing GstRTSPAddressPoolResult
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=714988
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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|
* gst/rtsp-server/rtsp-server.c:
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|
* gst/rtsp-server/rtsp-session-pool.c:
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|
* gst/rtsp-server/rtsp-session.c:
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* gst/rtsp-server/rtsp-stream.c:
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|
|
rtsp-server: sprinkle some allow-none annotations for g-i
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2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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stream: add method to filter transports
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|
Add a method to safely iterate and collect the stream transports
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
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2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-session-pool.c:
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* gst/rtsp-server/rtsp-session.c:
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rtsp: allow NULL func in filters
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Passing a null function make the filters return a list of
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refcounted objects.
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2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp-server/rtsp-address-pool.c:
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* tests/check/gst/addresspool.c:
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address-pool: fix address increment
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|
Use a guint instead of guint8 to increment the address. It's still not
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completely correct because a guint might not be able to hold the complete
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address range, but that's an enhacement for later.
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Add unit test to test improved behaviour.
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https://bugzilla.gnome.org/show_bug.cgi?id=708237
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2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* tests/check/gst/client.c:
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client: allow absolute path in requests
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
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2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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client: make make_path_from_uri a vmethod
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2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
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|
* docs/libs/gst-rtsp-server-sections.txt:
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* gst/rtsp-server/rtsp-stream.c:
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|
* gst/rtsp-server/rtsp-stream.h:
|
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|
* tests/check/Makefile.am:
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|
* tests/check/gst/stream.c:
|
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|
|
stream: Add functions to get rtp and rtcp sockets
|
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|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
|
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2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
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|
|
* gst/rtsp-server/rtsp-context.c:
|
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|
* gst/rtsp-server/rtsp-context.h:
|
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|
|
context: defing a GType for the context
|
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|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
|
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|
2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
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|
|
* gst/rtsp-server/Makefile.am:
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|
* gst/rtsp-server/rtsp-auth.c:
|
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|
* gst/rtsp-server/rtsp-context.c:
|
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|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
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|
|
* gst/rtsp-server/rtsp-server.h:
|
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|
|
* gst/rtsp-server/rtsp-session-media.c:
|
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|
* gst/rtsp-server/rtsp-session.c:
|
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|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
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|
|
Fixed several GIR warnings
|
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|
|
2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
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|
|
auth: small typos
|
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|
|
2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
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|
|
* tests/check/Makefile.am:
|
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|
|
* tests/check/gst/token.c:
|
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|
|
|
|
tests: Add unit tests for token
|
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|
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|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
|
|
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|
|
2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
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|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
token: Validate args for gst_rtsp_token_is_allowed
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=710520
|
|
|
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|
|
2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
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|
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|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
token: Fix bug when creating empty token
|
|
|
|
|
|
We always want to have a valid GstStructure in the token.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
|
|
|
|
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|
|
|
|
|
|
|
2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
thread-pool: avoid race in shutdown
|
|
|
|
|
|
If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
|
|
|
|
|
|
don't actually stop the mainloop ever. Solve this race by adding an idle source
|
|
|
|
|
|
to the mainloop that calls the _quit. This way we immediately exit the mainloop
|
|
|
|
|
|
if quit was called before we started it.
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/permissions.c:
|
|
|
|
|
|
tests: Add unit tests for permissions
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
tests: Test mediafactory permissions
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
permissions: Fix refcounting when adding/removing roles
|
|
|
|
|
|
Previously a role that was removed was unreffed twice, and when
|
|
|
|
|
|
replacing an existing role the replaced role was freed while still being
|
|
|
|
|
|
referenced. Both bugs are now fixed.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: Check gst_rtsp_url_parse return value
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
|
|
|
|
|
|
|
|
|
|
|
|
2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 865aa20 to dbedaa0
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: Fix socket leak
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=710088
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
rtsp-session-pool: Make sure session IDs are properly URI-escaped
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=643812
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
examples: fix compilation when WITH_AUTH is defined
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=710228
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
gitignore: Add new test binary
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/threadpool.c:
|
|
|
|
|
|
thread-pool: Add unit test for the thread pools
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=710228
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
thread-pool: Fix thread leak when reusing threads
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=709730
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: fixed racy behavior in rtspserver tests
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=710078
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
tests: Improve address pool unit tests
|
|
|
|
|
|
Add a range with mixed IPV4 and IPV6 addresses to pool.
|
|
|
|
|
|
Get an IPV4 address from an IPV6-only pool.
|
|
|
|
|
|
Get an IPV6 address from an IPV4-only pool.
|
|
|
|
|
|
Reserve a IPV6 address from an IPV4-only pool.
|
|
|
|
|
|
Check for unicast addresses in multicast-only pool.
|
|
|
|
|
|
Check for unicast addresses in uni-/multicast-mixed pool.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=710128
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: append query string in PAUSE/PLAY/TEARDOWN as well
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Add query to control path
|
|
|
|
|
|
If the SETUP url contains a query it must be appended to the control
|
|
|
|
|
|
path so that it matches any already created stream in the media. The
|
|
|
|
|
|
query will also be appended to the session media path.
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: remove old line
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: Correct control comparison
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=709176
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: Check dynamically if the pipeline supports seeking
|
|
|
|
|
|
We should not depend on whether or not the pipeline state change
|
|
|
|
|
|
returned NO_PREROLL or not. A media could dynamically change its
|
|
|
|
|
|
element and switch from seekable to non seekable so it's best to test
|
|
|
|
|
|
the seekable nature of the pipeline dynamically when we try to do a seek.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: Return FALSE if seeking is not supported
|
|
|
|
|
|
|
|
|
|
|
|
2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: don't seek accurate by default
|
|
|
|
|
|
Accurate seeking is perhaps a little overkill in the most common situation and
|
|
|
|
|
|
causes some formats (mp3) over slow media to seek extremely slowly.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: fix unit test
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Reply 400 if media cannot be constructed
|
|
|
|
|
|
Reply 400 Bad Request instead of 503 Service Unavailable if media
|
|
|
|
|
|
cannot be constructed in SETUP.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Send setup reply once only
|
|
|
|
|
|
If find_media() failed in handle_setup_request() two replies was sent.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6b03ba7 to 865aa20
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: Emit client-connected signal earlier
|
|
|
|
|
|
Emit client-connected before the client ref is given to a GSource,
|
|
|
|
|
|
otherwise client-connected can be emitted after the client object has
|
|
|
|
|
|
been freed.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
addresspool: return reason of failure
|
|
|
|
|
|
Let gst_rtsp_address_pool_reserve_address() return the reason why
|
|
|
|
|
|
the address could not be reserved.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
autogen.sh: Sync behaviour with other GStreamer modules
|
|
|
|
|
|
Allows building from outside of tree amongst other things
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From b613661 to 6b03ba7
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 74a6857 to b613661
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 01a7a46 to 74a6857
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Do not read beyond end of path string
|
|
|
|
|
|
If the setup was done without a control url, make sure we don't try to read the
|
|
|
|
|
|
non-existing control string and crash.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: Fix RTPInfo header
|
|
|
|
|
|
Refactor the method to make the content_base.
|
|
|
|
|
|
Use the content-base and the control url to construct the RTPInfo
|
|
|
|
|
|
url.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: map url to path only in describe
|
|
|
|
|
|
Only map the request url to a path in the DESCRIBE method. The SDP then
|
|
|
|
|
|
contains the base and control urls that should be used to SETUP/PAUSE/
|
|
|
|
|
|
PLAY/TEARDOWN the media.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Revert "client: map URL to path in requests"
|
|
|
|
|
|
This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
|
|
|
|
|
|
This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
|
|
|
|
|
|
contains the base and control urls which are used in the SETUP, PLAY,
|
|
|
|
|
|
PAUSE and TEARDOWN requests.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: map URL to path in requests
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
mount-points: make vmethod to make path from uri
|
|
|
|
|
|
Make a vmethod to transform an url into a path. The path is then used to lookup
|
|
|
|
|
|
the factory. This makes it possible to also use other bits of the url, such as
|
|
|
|
|
|
the query parameters, to locate the factory.
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
thread-pool: Add cleanup to wait for the threadpool to finish
|
|
|
|
|
|
Also fix race condition if two threads are asking for the first
|
|
|
|
|
|
thread from the thread pool at once. This would case two internal
|
|
|
|
|
|
GThreadPools to be created.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=707753
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
client: free threadpool
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=707638
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/mountpoints.c:
|
|
|
|
|
|
mountpoints tests: unref matched factories
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=707638
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media tests: unref thread pool and caps
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=707638
|
|
|
|
|
|
|
|
|
|
|
|
2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
auth, media, media-factory: unref permissions
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=707638
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
Makefile: add rule for appsrc example
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-appsrc.c:
|
|
|
|
|
|
tests: add appsrc example
|
|
|
|
|
|
Add an example on how to use appsrc to feed the server pipeline with data.
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: remove query part from content-base string
|
|
|
|
|
|
Make sure that after the control url has been resolved, it's
|
|
|
|
|
|
not a part of the query-string.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: don't check url in response
|
|
|
|
|
|
There is no url or method in the response to check
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Add handle-response signal for when we receive a GET_PARAMETER response
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
Fix gst_rtsp_server_client_filter, using wrong variable type
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
|
|
|
|
|
|
For AAC we need to check for framed=true instead of parsed=true.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=701384
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: optimize pipeline for protocols
|
|
|
|
|
|
When TCP is not an allowed protocol for the stream, avoid creating the
|
|
|
|
|
|
appsrc/appsink/queue and tee elements.
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: set protocols on streams
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use protocols supported by stream
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
media-factory: allow all protocols
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: configure protocols in new streams
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add protocols property
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: send state in "new-state" signal
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=705110
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
build: add subdir-objects to AM_INIT_AUTOMAKE
|
|
|
|
|
|
Fixes warnings with automake 1.14
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=705350
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: add method to iterate clients of server
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add vmethod for rtsp-media subclass to access rtpbin
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
small documentation fix
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Do not take range header if range is invalid
|
|
|
|
|
|
|
|
|
|
|
|
2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: add docs for new method
|
|
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|
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|
|
2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
* gst/rtsp-server/rtsp-media.h:
|
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|
|
Add API to rtsp-media set the pipeline's state
|
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|
2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
Update current position/duration when gst_rtsp_media_get_range_string is called
|
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|
|
2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-cgroups.c:
|
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|
|
tests: add some more docs
|
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|
|
2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-cgroups.c:
|
|
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|
|
* gst/rtsp-server/Makefile.am:
|
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
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|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
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|
|
* gst/rtsp-server/rtsp-client.h:
|
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|
|
|
|
* gst/rtsp-server/rtsp-context.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-context.h:
|
|
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|
|
|
* gst/rtsp-server/rtsp-params.c:
|
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|
|
* gst/rtsp-server/rtsp-params.h:
|
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|
|
* gst/rtsp-server/rtsp-server.c:
|
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|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
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|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
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|
|
* tests/check/gst/client.c:
|
|
|
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|
|
ClientState -> Context
|
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|
|
Rename the clientstate to context and put the code in a separate file.
|
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|
|
2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-auth.c:
|
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
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|
|
* gst/rtsp-server/rtsp-auth.h:
|
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|
|
auth: add support for default token
|
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|
|
The default token is used when the user is not authenticated and can be used to
|
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|
|
give minimal permissions.
|
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|
2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-auth.c:
|
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
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|
|
|
auth: use defines when possible
|
|
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|
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|
|
2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
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|
|
address-pool: improve docs
|
|
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|
2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-permissions.c:
|
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|
|
permissions: add the role to the copy
|
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|
2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
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|
|
permissions: Also copy the roles
|
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|
|
2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
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|
|
permissions: Make it build
|
|
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|
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|
|
2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
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|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
docs: small fixes
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
docs: improve docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
address-pool: cleanups
|
|
|
|
|
|
Remove redundant method, improve docs.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
docs: improve docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
permissions: implement _remove_role
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
permissions: update docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: simplify tests
|
|
|
|
|
|
Client settings are now disabled by default so we don't need an auth
|
|
|
|
|
|
module to disable them.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: add default authorizations
|
|
|
|
|
|
When no auth module is specified, use our table of defaults to look up the
|
|
|
|
|
|
default value of the check instead of always allowing everything. This was
|
|
|
|
|
|
we can disallow client settings by default.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
README: update readme
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
thread-pool: add more docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
thread-pool: fix race in thread reuse
|
|
|
|
|
|
If we try to reuse a thread right after we made it stop, we end up using a
|
|
|
|
|
|
stopped thread. Catch this case and only reuse threads that are not stopping.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: add small debug
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
client: fix test
|
|
|
|
|
|
Add some permissions to media so we can use the auth and enable
|
|
|
|
|
|
client settings.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: support pushed context in handle_request
|
|
|
|
|
|
If we already have a pushed state, reuse it and add our own things. This makes
|
|
|
|
|
|
it easier to write tests.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: don't auth on methods
|
|
|
|
|
|
Don't authorize on methods anymore but on the resources that we
|
|
|
|
|
|
try to access, this is more flexible.
|
|
|
|
|
|
Move the authorization checks to where they are needed and let the
|
|
|
|
|
|
check return the response on error.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
mount-points: add some debug
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: almost fix test
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
auth: let the auth module check client_settings
|
|
|
|
|
|
Let the auth module decide if client settings are allowed for the
|
|
|
|
|
|
current client.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.h:
|
|
|
|
|
|
token: add method to check boolean permission
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-token.h:
|
|
|
|
|
|
token: simplify token constructor
|
|
|
|
|
|
Use variable arguments to make easier API.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add convenience API for factory
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
permissions: simplify API a little
|
|
|
|
|
|
Avoid passing GstStructure in the add_role method, use varargs instead
|
|
|
|
|
|
to construct the structure behind the scenes. We can then also use the
|
|
|
|
|
|
structure name as the role and simplify some more logic.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: fix typo
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
auth: handle unauthorized response
|
|
|
|
|
|
Move handling of the unauthorized response to the auth module, it can add
|
|
|
|
|
|
the appropriate headers to request authorization for the required method
|
|
|
|
|
|
much better than the client.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: allow for sending any message, not only requests
|
|
|
|
|
|
Change the _send_request() method to _send_message() so that we
|
|
|
|
|
|
can both send requests and replies.
|
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|
|
2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
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|
|
|
docs: fix docs
|
|
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|
|
|
|
2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
auth: move TLS handling to auth module
|
|
|
|
|
|
Remove the TLS settings on the server and move it to the auth module because
|
|
|
|
|
|
that is where security related bits go.
|
|
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|
|
2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add state push/pop
|
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|
|
2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
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|
|
|
* gst/rtsp-server/rtsp-client.h:
|
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|
|
client: add connection to state
|
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|
|
2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
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|
|
mount-points: fix debug
|
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|
|
2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
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|
|
* tests/check/gst/media.c:
|
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|
|
tests: fix media test
|
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|
|
2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
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|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
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|
|
|
thread-pool: we don't require a state
|
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|
|
2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
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|
|
* gst/rtsp-server/rtsp-server.c:
|
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|
|
server: let context ref the server
|
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|
|
So that we don't risk losing the server object early anc crash.
|
|
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|
|
2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
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|
|
|
|
* tests/check/gst/client.c:
|
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|
|
tests: fix client test
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.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-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.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
docs: improve docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
session-pool: make vmethod to create a session
|
|
|
|
|
|
Make a vmethod to create a sessions so that subclasses can create
|
|
|
|
|
|
custom session objects
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
docs: more updates
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
docs: update docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
configure: compile cgroup example conditionally
|
|
|
|
|
|
Only compile the cgroup example when we have libcgroup
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-cgroups.c:
|
|
|
|
|
|
examples: add cgroups example
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: fix compilation
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
thread-pool: fix vmethod invocation
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
thread-pool: store thread type in thread
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: pass thread from pool to media _prepare
|
|
|
|
|
|
Get a thread from the configured threadpool and pass it to the prepare method of
|
|
|
|
|
|
the media.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: Accept a thread in _prepare
|
|
|
|
|
|
Remove out own threadpool handling and use the provided thread and
|
|
|
|
|
|
maincontext for the bus messages and the state changes.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: configure client thread pool
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add method to configure thread pool
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: use thread pool
|
|
|
|
|
|
Use the thread pool instead of doing our own thing.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-thread-pool.h:
|
|
|
|
|
|
thread-pool: add object to manage threads
|
|
|
|
|
|
Add an object to manage the client and media threads.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: debug authorization check
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: start media pipeline in context
|
|
|
|
|
|
Start the media pipeline in the provided context (or our default one
|
|
|
|
|
|
when NULL). This makes sure that we run the bus thread in this context and that
|
|
|
|
|
|
all media threads are children of this context.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
factory: pass permissions to media by default
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
test: add permissions to auth test
|
|
|
|
|
|
Ass some permissions to the media factory in the test.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
auth: simplify auth checks
|
|
|
|
|
|
Remove client from methods, it's now in the state
|
|
|
|
|
|
Perform the check specified by the string, use the information from the
|
|
|
|
|
|
thread local context.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add state to current thread
|
|
|
|
|
|
Add the client to the ClientState object.
|
|
|
|
|
|
Place the ClientState on the current thread.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
media: make it possible to set permissions
|
|
|
|
|
|
Make it possible to set permissions on media and media factory objects
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-permissions.h:
|
|
|
|
|
|
permissions: add permissions object
|
|
|
|
|
|
Add a mini object to store permissions based on a role.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
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* examples/test-auth.c:
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
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|
* gst/rtsp-server/rtsp-auth.h:
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
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auth: add auth checks
|
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|
|
Add an enum with auth checks and implement the checks in the auth object.
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Perform the checks from the client.
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2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* examples/test-auth.c:
|
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* gst/rtsp-server/rtsp-auth.c:
|
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* gst/rtsp-server/rtsp-auth.h:
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* gst/rtsp-server/rtsp-client.h:
|
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auth: use the token after authentication
|
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After we authenticated a user, keep the Token around in the state.
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2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
|
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* tests/check/gst/media.c:
|
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media: add optional context for bus messages
|
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|
Add an optional mainloop to _prepare that will handle the bus messages instead
|
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|
of always using the shared mainloop.
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2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/Makefile.am:
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* gst/rtsp-server/rtsp-token.c:
|
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* gst/rtsp-server/rtsp-token.h:
|
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|
|
token: add authorization token
|
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|
|
Add a simply miniobject that contains the authorizations. The object contains a
|
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|
|
GstStructure that hold all authorization fields. When a user is authenticated,
|
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|
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the auth module will create a Token for the user. The token is then used to
|
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|
check what operations the user is allowed to do and various other configuration
|
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values.
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2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* examples/test-auth.c:
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
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* gst/rtsp-server/rtsp-auth.h:
|
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|
* gst/rtsp-server/rtsp-client.c:
|
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* gst/rtsp-server/rtsp-client.h:
|
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|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
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|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
* gst/rtsp-server/rtsp-media.h:
|
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|
|
auth: remove auth from media and factory
|
|
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|
|
Remove the auth object from media and factory. We want to have the RTSPClient
|
|
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|
|
|
authenticate and authorize resources, there is no need to place another auth
|
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|
|
|
manager on the media/factory.
|
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2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
|
|
* examples/test-auth.c:
|
|
|
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|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
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|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
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|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
auth: add support for multiple basic auth tokens
|
|
|
|
|
|
Make it possible to add multiple basic authorisation tokens to one authorization
|
|
|
|
|
|
object. Associate with each token an authorization group that will define what
|
|
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|
|
|
capabilities are allowed.
|
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|
|
2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
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|
|
client: error out on non-aggregate control
|
|
|
|
|
|
We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
|
|
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|
|
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|
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|
|
2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: rework setup request a little
|
|
|
|
|
|
Cache the media in DESCRIBE based on the longest matching path with the uri
|
|
|
|
|
|
that we can find in the mount points.
|
|
|
|
|
|
Rework the setup request a little to get the media from the session or from
|
|
|
|
|
|
the longest matching path, this way we can derive the control string as
|
|
|
|
|
|
everything after the path instead of hardcoding it.
|
|
|
|
|
|
Find the stream based on the control string and only open a session when all
|
|
|
|
|
|
this can be done.
|
|
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|
|
2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add method to find a stream by control url
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add method to check control url of stream
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
session: use path matching for session media
|
|
|
|
|
|
Use a path string instead of a uri to lookup session media in the sessions. Also
|
|
|
|
|
|
use path matching to find the largest possible path that matches.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* tests/check/gst/mountpoints.c:
|
|
|
|
|
|
mount-points: remove useless vmethod
|
|
|
|
|
|
Making lookups in the mount points should not be done with a URL, if there is a
|
|
|
|
|
|
mapping to be done from URL to mount points, we'll need to do it somewhere
|
|
|
|
|
|
else.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* tests/check/gst/mountpoints.c:
|
|
|
|
|
|
mount-points: improve mount point searching
|
|
|
|
|
|
Use a GSequence to keep track of the mount points.
|
|
|
|
|
|
Match a URL to the longest matching registered mount point. This should be the
|
|
|
|
|
|
URL to perform aggreagate control and the remainder is the stream specific
|
|
|
|
|
|
control part.
|
|
|
|
|
|
Add some unit tests for this.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
rtsp-server: Allow building of static library
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
tests: fix compilation
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: get control string from stream
|
|
|
|
|
|
Use the control string as configured in the stream.
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add methods and property to set control string
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: cleanups
|
|
|
|
|
|
Rename variables for clarity
|
|
|
|
|
|
Keep media in state when we can
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add more support for IPv6
|
|
|
|
|
|
Rename _get_address to _get_multicast_address in GstRTSPStream to
|
|
|
|
|
|
make it clear that this function only deals with multicast.
|
|
|
|
|
|
Make it possible to have both an IPv4 and IPv6 multicast address on
|
|
|
|
|
|
a stream. Give the client an IPv4 or IPv6 address depending on the
|
|
|
|
|
|
address it used to connect to the server.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix comment
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: handle failed port allocation
|
|
|
|
|
|
Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
|
|
|
|
|
|
can't allocate any family at all. Also keep track of what port families we
|
|
|
|
|
|
allocated.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: improve docs
|
|
|
|
|
|
|
|
|
|
|
|
2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
stream-transport: remove old if 0 block
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: fix tests
|
|
|
|
|
|
gst_rtsp_client_get_uri() has been removed
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add method to filter managed sessions
|
|
|
|
|
|
Add a method to filter the sessions managed by this client connection.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=703016
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: remove _get_uri() method
|
|
|
|
|
|
Remove the get_uri() method on the client. A client has no uri, the uri
|
|
|
|
|
|
property is an internal property to manage the last cached media for
|
|
|
|
|
|
the client.
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: fix typo
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Do not leak the query in default_query_stop
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't unlock when conversion fails
|
|
|
|
|
|
Don't unlock the state lock when conversion fails because it was not locked.
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add query_position and query_stop vmethods to rtsp-media
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Fix typo in property install for rtsp-media's time-provider
|
|
|
|
|
|
|
|
|
|
|
|
2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: clean some variables
|
|
|
|
|
|
Clean some variables and add some guards to _send_request()
|
|
|
|
|
|
|
|
|
|
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2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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Add gst_rtsp_client_send_request API
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This makes it possible to send arbitrary messages to a client, such as
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SET_PARAMETER or GET_PARAMETER
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2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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media: add _get_element() method
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Add method to get the element used when creating the media.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
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2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media.c:
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media: fix docs
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2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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stream: allow access to the rtp session
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https://bugzilla.gnome.org/show_bug.cgi?id=703004
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2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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dscp qos support in gst-rtsp-stream
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
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2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* tests/check/gst/rtspserver.c:
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tests: fix test
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Actually do what the comment says. Also keep the old code around, not sure what
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should happen when you get a 454 from a TEARDOWN, does it close the connection?
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it currently doesn't.
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2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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client: also watch newly created session
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When we newly created a session, start watching it immediately instead of
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on the next request.
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2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
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* tests/check/gst/client.c:
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tests: add unit test for new-session
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See https://bugzilla.gnome.org/show_bug.cgi?id=701587
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2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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client: emit new-session when new session is created
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Only emit new-session when we created a new session for a client, not when a
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client picked up a previous session.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
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2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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client: handle asterisk as path in requests
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
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2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media.c:
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media: handle segment query format mismatch
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It's possible that the segment query returns with a different format than what
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we asked for, handle this case also.
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2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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media: use segment stop in collect_media_stats
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Use segment stop instead of duration as range end point.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
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2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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* tests/check/gst/media.c:
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rtsp-media: Do not leak the element in take_pipeline
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
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2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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rtsp-client: Make configure_client_transport virtual
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This patch makes configure_client_transport virtual. The functionality is
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needed to handle some weird clients sending multicast transport settings as url
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options.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
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2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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rtsp-client: Make param_set and param_get virtual
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
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2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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media: convert_range replaces get_range_times
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get_range_times worked for handling UTC ranges for seeks, but we also
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need to convert back from NPT to the requested unit in
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get_range_string. convert_range is now used for both.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
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2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-sdp.c:
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* gst/rtsp-server/rtsp-sdp.h:
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sdp: cleanup sdp info
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We don't need to pass the proto, we can more easily check a boolean.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
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2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
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* gst/rtsp-server/rtsp-sdp.c:
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use 0.0.0.0 or :: for c= line instead of server address
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2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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use local address, not remote, in SDP
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See https://bugzilla.gnome.org/show_bug.cgi?id=702063
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2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* common:
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Automatic update of common submodule
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From 098c0d7 to 01a7a46
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2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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media: possibility to override range time conversion
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Make it possible to override the conversion from GstRTSPTimeRange to
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GstClockTimes, that is done before seeking on the media
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pipeline. Overriding can be useful for UTC ranges, where the default
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conversion gives nanoseconds since 1900.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
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2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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rtsp-server: Expose the use_client_settings API
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
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2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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rtspstream: handle both ipv4 and ipv6 clients
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
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2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-sdp.c:
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Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
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This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
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We already have a way to place extra attributes in the SDP by using a string
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property with prefix x- or a- in the caps.
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2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-sdp.c:
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Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
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This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
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We already have a way to place extra attributes in the SDP, just make a string
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property in the payloader with a- or x- prefix.
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2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-sdp.c:
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rtsp: place a- and x- properties as attributes
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application/x-rtp has properties with a- and x- prefixes that should be
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placed as attributes in the SDP for the media instead of being added to the
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fmtp.
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2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* examples/Makefile.am:
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* examples/test-video.c:
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example: add TLS example
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2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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server: add support for TLS
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Add methods to set and get a TLS certificate.
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Add vmethod to configure a new connection. By default, configure the TLS
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certificate in a new connection if needed.
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2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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server: remove accept_client vmethod
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This vmethod is not very useful so remove it.
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2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-server.c:
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server: don't crash on NULL GError
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|
2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-session-pool.c:
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rtsp-session-pool: corrected session timeout detection
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|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
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|
2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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client: improve debug
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2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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|
* gst/rtsp-server/rtsp-server.c:
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|
server: refactor connection setup
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|
|
Let the server accept the socket connection and construct a GstRTSPConnection
|
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|
from it. Remove the code from the client and let the client only deal with
|
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|
a fully configure GstRTSPConnection object.
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We will need this later when the server will configure the connection for
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TLS.
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|
2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
* gst/rtsp-server/rtsp-stream.c:
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|
stream: keep the transport object alive
|
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|
|
Keep the transport object alive while we have it as qdata on the
|
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source.
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|
2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
* gst/rtsp-server/rtsp-server.c:
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|
|
rtsp-server: Do not crash on nmapping of server
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|
|
* generate error when gst_rtsp_connection_accept fails
|
|
|
|
|
|
* do not stop accepting incoming connections because
|
|
|
|
|
|
accepting a client fails
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=701072
|
|
|
|
|
|
|
|
|
|
|
|
2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=700953
|
|
|
|
|
|
|
|
|
|
|
|
2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-sdp: Parse framerate caps field and set SDP attribute
|
|
|
|
|
|
The SDP attribute and its format is described in RFC4566.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
|
|
|
|
|
|
|
|
|
|
|
|
2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-sdp: Parse width/height from caps and set SDP attribute
|
|
|
|
|
|
The SDP attribute and its format is described in RFC6064.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
rtsp-sdp: add bandwidth line
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=699220
|
|
|
|
|
|
|
|
|
|
|
|
2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 5edcd85 to 098c0d7
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
tests: add dynamic payloader prepare/unprepare check
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: release lock when removing fakesink
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: set elements to NULL before removing
|
|
|
|
|
|
When removing a stream, set the elements to NULL first. This avoids
|
|
|
|
|
|
element-is-not-in-NULL-state errors when we dispose the elements.
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 3cb3d3c to 5edcd85
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: listen to pad-removed signals
|
|
|
|
|
|
Listen to the pad-removed signal and remove the stream associated with the
|
|
|
|
|
|
removed pad.
|
|
|
|
|
|
Add signal to be notified of the removed pad.
|
|
|
|
|
|
Remove the fakesink in unprepare()
|
|
|
|
|
|
Fix signatures of the signal methods
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
tests: add example of reusable pipelines
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add method to get the srcpad
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
check: add media prepare/unprepare test
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: disconnect from signal handlers in unprepare()
|
|
|
|
|
|
We connected to the pad-added and no-more-pads signals in prepare() so
|
|
|
|
|
|
we need to disconnect from them in unprepare().
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't free streams array
|
|
|
|
|
|
Don't free the streams array in the unprepare() method, they were not
|
|
|
|
|
|
added in prepare().
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't unref the pipeline in unprepare
|
|
|
|
|
|
Unprepare() should undo what prepare() does. Because the pipeline is
|
|
|
|
|
|
not created in prepare(), we should not unref it in unprepare()
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: clear session and caps for reuse
|
|
|
|
|
|
Set the session and caps to NULL after unref otherwise we might unref
|
|
|
|
|
|
them again later.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: send out teardown signal before tearing down
|
|
|
|
|
|
The advantage is that in the signal handler you get direct access to
|
|
|
|
|
|
information about what streams are about to get torn down (in the
|
|
|
|
|
|
GstRTSPClientState).
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: expose connection
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From aed87ae to 3cb3d3c
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
media: add method to get the base_time of the pipeline
|
|
|
|
|
|
Together with a shared clock, this base-time could eventually be sent to
|
|
|
|
|
|
the client so that it can reconstruct the exact running-time of the clock
|
|
|
|
|
|
on the server.
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
media: add GstNetTimeProvider support
|
|
|
|
|
|
Add a property to let the media provide a GstNetTimeProvider for its clock.
|
|
|
|
|
|
Make methods to get the clock and nettimeprovider
|
|
|
|
|
|
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
|
|
|
|
|
|
provider and also the current time of the clock. This should make it possible
|
|
|
|
|
|
for (GStreamer) clients to slave their clock to the server clock.
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 04c7a1e to aed87ae
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: wait for buffering to complete
|
|
|
|
|
|
Wait for buffering to complete before changing the state to the target state.
|
|
|
|
|
|
|
|
|
|
|
|
2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: small cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: remove extra unref in test_setup_non_existing_stream
|
|
|
|
|
|
The unref is not needed anymore, teardown runs without it.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=696542
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: GSocketService cleanup in test_bind_already_in_use
|
|
|
|
|
|
Use g_socket_service_stop so the rtspserver test stops listening for
|
|
|
|
|
|
incoming connections in test_bind_already_in_use.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=696541
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
|
|
|
|
|
|
Instead use a GWeakRef which is safe to use
|
|
|
|
|
|
This is a known GLib bug, see:
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=667145
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtsp-media/client: Reply to PLAY request with same type of Range
|
|
|
|
|
|
Remember the type of Range from the PLAY request and use the same type for
|
|
|
|
|
|
the reply.
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
rtsp-client: expose uri
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
tests: Hold ref while creating second media
|
|
|
|
|
|
To test if the media aren't shared, make sure we keep the first one while creating a second
|
|
|
|
|
|
otherwise the same memory address may be reused.
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: remove out-of-date comment
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
.gitignore: ignore more build files
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: use right _LIBS variable for gst-plugins-base libs
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
check: add librtp to libs
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: Add test to check selecting a port the server will send from
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
tests: Make sure packets are actually received
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: Select unicast address from pool if appropriate
|
|
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|
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|
|
2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
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|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: Properties are always there in Gst 1.0
|
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|
2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
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|
|
* tests/check/gst/addresspool.c:
|
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|
|
|
tests: Add tests for unicast addresses in pool
|
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|
2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
address-pool: Verify that multicast addresses are used for multicast and vice-versa
|
|
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|
2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
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|
|
|
address-pool: Add unicast addresses
|
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2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
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|
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* configure.ac:
|
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|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtsp-server: Limit the number of threads per server instance
|
|
|
|
|
|
If we exceed the maximum, just round robin the clients over the existing
|
|
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|
|
threads.
|
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2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
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|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: No need to store the GMainContext in the client context
|
|
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|
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|
2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
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|
|
* tests/check/gst/rtspserver.c:
|
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|
|
|
tests: Add test for client disconnection
|
|
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|
2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
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|
|
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|
|
* tests/check/gst/rtspserver.c:
|
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|
|
|
tests: Test client and session timeouts with multiple threads
|
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|
2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
|
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|
|
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|
|
|
* 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:
|
|
|
|
|
|
Document locking and its order
|
|
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|
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|
2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
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|
|
|
tests: Test that slow DESCRIBE don't block other clients
|
|
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|
|
2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
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|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: Add tests for client-requested multicast address
|
|
|
|
|
|
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|
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|
|
2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
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|
|
|
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|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
docs: Put the various functions in the right sections
|
|
|
|
|
|
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|
|
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|
|
2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
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|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
docs: Generate docs for GstRTSPAddressPool
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
client: Check client provided addresses against the address pool
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
address-pool: Add API to request a specific address from the pool
|
|
|
|
|
|
Also add relevant unit tests.
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/mediafactory.c:
|
|
|
|
|
|
tests: Check the passing around of a RTSPAddressPool
|
|
|
|
|
|
Make sure the RTSPAddressPool is propagated from the MediaFactory all the
|
|
|
|
|
|
way down to the stream.
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
tests: Add more tests for the address pool
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
address-pool: Fix off by one error
|
|
|
|
|
|
When splitting a port range, the port after a skip is not part of range.
|
|
|
|
|
|
|
|
|
|
|
|
2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 2de221c to 04c7a1e
|
|
|
|
|
|
|
|
|
|
|
|
2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
|
|
|
|
|
|
AM_CONFIG_HEADER was removed in automake 1.13
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=693368
|
|
|
|
|
|
|
|
|
|
|
|
2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From a942293 to 2de221c
|
|
|
|
|
|
|
|
|
|
|
|
2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: make sure the watch exists while sending data
|
|
|
|
|
|
Protect the send_func with a lock. This allows us to wait for sending
|
|
|
|
|
|
to complete before changing the send_func and user_data. We add an
|
|
|
|
|
|
extra ref to the watch to make sure that it remains valid during
|
|
|
|
|
|
sending.
|
|
|
|
|
|
When closing the connection, set the send_func to NULL
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
|
|
|
|
|
|
|
|
|
|
|
|
2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: use GST_*_1_0 environment variables everywhere
|
|
|
|
|
|
The _1_0 suffixed environment variables override the
|
|
|
|
|
|
non-suffixed ones, so if we're in an environment that
|
|
|
|
|
|
sets the _1_0 suffixed ones, such as jhbuild, we need
|
|
|
|
|
|
to set those to make sure ours actually always get
|
|
|
|
|
|
used.
|
|
|
|
|
|
|
|
|
|
|
|
2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From acb04d9 to a942293
|
|
|
|
|
|
|
|
|
|
|
|
2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: set the client backlog
|
|
|
|
|
|
Set the client backlog to a reasonable default
|
|
|
|
|
|
|
|
|
|
|
|
2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: Make the element a constructor parameter
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=689594
|
|
|
|
|
|
|
|
|
|
|
|
2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
docs: Link with gcov library when gcov is enabled
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: match prepare with unprepare
|
|
|
|
|
|
Really unprepare when there were an equal amount of prepare calls.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: media has to be unprepared in finalize
|
|
|
|
|
|
Because unprepare takes away the last ref on the media.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
|
|
|
|
|
|
This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
|
|
|
|
|
|
We can't use the refcount to trigger unprepare because it is the unprepare call
|
|
|
|
|
|
that removes the last refcount after all messages are consumed. What we should
|
|
|
|
|
|
probably do is make a prepared refcount and only unprepare when the refcount
|
|
|
|
|
|
reaches 0.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: let the source unref the last media ref
|
|
|
|
|
|
the last ref to the media is held by the source so we don't need to add more ref
|
|
|
|
|
|
and unrefs, we simply destroy the media when the source is gone.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: improve debug
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: check state
|
|
|
|
|
|
Make sure we are in the right state when collecting the position and duration.
|
|
|
|
|
|
Only make ourselves PREPARED when we were previously PREPARING.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: use g_object_ref/unref for GObjects
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
|
|
|
|
|
|
Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
|
|
|
|
|
|
GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
|
|
|
|
|
|
isn't being used anymore.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Fix compiler warning
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
small cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* tests/check/gst/media.c:
|
|
|
|
|
|
media: avoid element leak
|
|
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|
2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
media: require an element in media constructor
|
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|
|
2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
Revert "client: TEARDOWN brings that state to Init again"
|
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|
|
This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
|
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The object is already disposed, there is no point in setting the state.
|
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2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
|
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|
|
client: TEARDOWN brings that state to Init again
|
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2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* docs/libs/gst-rtsp-server-sections.txt:
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* examples/test-auth.c:
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* gst/rtsp-server/rtsp-auth.c:
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* gst/rtsp-server/rtsp-auth.h:
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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* gst/rtsp-server/rtsp-media-factory-uri.c:
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* gst/rtsp-server/rtsp-media-factory-uri.h:
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media-factory.h:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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* gst/rtsp-server/rtsp-mount-points.c:
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* gst/rtsp-server/rtsp-mount-points.h:
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* gst/rtsp-server/rtsp-sdp.c:
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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* gst/rtsp-server/rtsp-session-media.c:
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* gst/rtsp-server/rtsp-session-media.h:
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* gst/rtsp-server/rtsp-session-pool.c:
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* gst/rtsp-server/rtsp-session-pool.h:
|
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* gst/rtsp-server/rtsp-session.c:
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* gst/rtsp-server/rtsp-session.h:
|
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|
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* gst/rtsp-server/rtsp-stream-transport.c:
|
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* gst/rtsp-server/rtsp-stream-transport.h:
|
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* gst/rtsp-server/rtsp-stream.c:
|
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* gst/rtsp-server/rtsp-stream.h:
|
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* tests/check/gst/media.c:
|
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|
|
rtsp: make object details private
|
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Make all object details private
|
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|
Add methods to access private bits
|
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|
2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* tests/check/Makefile.am:
|
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|
* tests/check/gst/media.c:
|
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|
|
tests: add media tests
|
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|
2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
media: check if prepared for some methods
|
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|
|
Check that the media object is prepared before doing seek and getting the
|
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|
current position etc.
|
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|
|
Add some g_return checks.
|
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|
|
2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
* tests/check/Makefile.am:
|
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|
|
* tests/check/gst/mediafactory.c:
|
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|
|
|
tests: add mediafactory test
|
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|
|
2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
* gst/rtsp-server/rtsp-stream.c:
|
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|
|
stream: improve debug
|
|
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|
|
2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
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|
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|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: unref pipeline in finalize to avoid leaking it
|
|
|
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|
|
2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp: use gst_object_unref on GstObjects
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: require an url
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: fix include
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: remove unused include
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/mountpoints.c:
|
|
|
|
|
|
tests: add test for mountpoints
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix factory leak
|
|
|
|
|
|
Keep the factory in the state object only for authorization checks and make
|
|
|
|
|
|
sure we unref it on failure. Also don't keep invalid objects in the state
|
|
|
|
|
|
object.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
mounts: add g_return_if guards
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: add more tests
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: improve debug
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: improve debug and fix leaks
|
|
|
|
|
|
Cleanup the uri and session when there is a bad request.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
update common
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
test: add test for session in options request
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use 454 when session can't be found
|
|
|
|
|
|
We should use 454 when a session can't be found because there was no session
|
|
|
|
|
|
pool configured in the server. This is not a server configuration problem
|
|
|
|
|
|
because the server on which the request is done might not be the same one that
|
|
|
|
|
|
will keep the sessions for us and so it does not need to support sessions.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: only free connection when there is one
|
|
|
|
|
|
It's possible that the client doesn't have a connection when we try to free it.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/client.c:
|
|
|
|
|
|
tests: add unit test for the client object
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: small cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: remove unused include
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix compilation
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: call destroy without the lock
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: make the client usable without a socket
|
|
|
|
|
|
Make a method to let the client handle a message and a callback when the client
|
|
|
|
|
|
wants us to send a response message back. This makes it possible to also use the
|
|
|
|
|
|
client object without the sockets, which should make it easier to test.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: small cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
client: remove reference to server
|
|
|
|
|
|
We don't need to keep a ref to the server
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add locking
|
|
|
|
|
|
Also add some g_return_if()
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: log more errors
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix compilation
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add generic close-after-send support
|
|
|
|
|
|
Add a property to send_response() to close the connection after the response has
|
|
|
|
|
|
been sent to the client.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server.types:
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-multicast.c:
|
|
|
|
|
|
* examples/test-multicast2.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
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|
|
* gst/rtsp-server/rtsp-mount-points.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-mount-points.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
MediaMapping -> MountPoints
|
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|
|
Describes better what the object manages.
|
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|
|
2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
|
|
* configure.ac:
|
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|
|
|
|
configure: bump required version of -base
|
|
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|
|
2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix seeking
|
|
|
|
|
|
|
|
|
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|
|
2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: support more Range formats
|
|
|
|
|
|
Use the new -base methods to convert the Range string into a seek start and stop
|
|
|
|
|
|
value.
|
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|
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|
|
2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
examples: fix whitespace
|
|
|
|
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|
|
|
|
|
|
|
2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
test-auth: add example of how to remove sessions
|
|
|
|
|
|
Add an example of the session filter api.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
test-uri: remove mapping example
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
test-uri: fix callback signature
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
factory: keep ref to factory while media active
|
|
|
|
|
|
While the media from a factory is alive, keep a ref to the factory.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
factory-uri: add some debug
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: set udp sources to PLAYING
|
|
|
|
|
|
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
|
|
|
|
|
|
so that it doesn't cause our pipeline to produce ASYNC-DONE.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
factory-uri: take ref to factory
|
|
|
|
|
|
Take a ref to the factory that we place in our list.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/Makefile.am:
|
|
|
|
|
|
* tests/test-reuse.c:
|
|
|
|
|
|
test: add test for server reuse
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=688395
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: start and stop multiple times
|
|
|
|
|
|
Stop listening on the RTSP port when the GSource is removed, so clients
|
|
|
|
|
|
can't connect and the server can be started again.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: fix small leak
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: unref source in finish_unprepare
|
|
|
|
|
|
The source is created in prepare, unref it in finish_unprepare.
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=688707
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: remove bus watch before finalizing
|
|
|
|
|
|
* A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
|
|
|
|
|
|
* An extra media ref is added for the bus watch. This extra ref is unreffed by
|
|
|
|
|
|
the GDestroyNotify function.
|
|
|
|
|
|
* gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
|
|
|
|
|
|
* GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
|
|
|
|
|
|
gst_rtsp_media_unprepare before unreffing the media.
|
|
|
|
|
|
This way, the bus watch will be removed before the media is finalized.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: wait until the TEARDOWN response is sent to close the connection
|
|
|
|
|
|
Responses can be sent async so we need to wait until the TEARDOWN response has
|
|
|
|
|
|
been written before we close the connection to the client. This avoids the risk
|
|
|
|
|
|
of writing/polling closed sockets.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp-stream: plug socket leak
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6bb6951 to a72faea
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
rtsp-server: don't use deprecated API
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: fix unused-but-set-variable compiler warning
|
|
|
|
|
|
rtsp-client.c:1260:21: error: variable 'protocols' set but not used
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* TODO:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp: cleanups
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-multicast2.c:
|
|
|
|
|
|
examples: add another multicast example
|
|
|
|
|
|
Add an example for how to configure separate multicast ranges for each media
|
|
|
|
|
|
stream.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-multicast.c:
|
|
|
|
|
|
test: set shared
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
stream: use the address managed by the stream
|
|
|
|
|
|
Use the address managed by the stream for multicast. This allows us to have 1
|
|
|
|
|
|
multicast address for each stream.
|
|
|
|
|
|
Because the address is now managed by the stream we don't have to pass it around
|
|
|
|
|
|
anymore.
|
|
|
|
|
|
Set the address pool on the streams.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
rtsp: improve debug
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add signal for new streams
|
|
|
|
|
|
This allows applications to listen for new streams and configure properties on
|
|
|
|
|
|
them, like the address pool.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: configure address pool in new streams
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add methods to deal with address pool
|
|
|
|
|
|
Add methods to get and set the address pool for the stream
|
|
|
|
|
|
Add method to allocate and get the multicast addresses for this stream.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: remove MTU property
|
|
|
|
|
|
It is a stream property
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: set blocksize only on stream
|
|
|
|
|
|
Set the blocksize only on the current stream.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream: share src and sink sockets
|
|
|
|
|
|
the allocated socket is in the used-socket property, not socket.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
rtsp: make address-pool return an address object
|
|
|
|
|
|
Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
|
|
|
|
|
|
store more info in the structure and allows us to more easily return the address
|
|
|
|
|
|
to the right pool when no longer needed.
|
|
|
|
|
|
Pass the address to the StreamTransport so that we can return it to the pool
|
|
|
|
|
|
when the stream transport is freed or changed.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-multicast.c:
|
|
|
|
|
|
examples: add multicast example
|
|
|
|
|
|
Show how to set up the multicast address pool so that media can be
|
|
|
|
|
|
server with multicast.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
rtsp: use AddressPool
|
|
|
|
|
|
Remove the multicast_group property.
|
|
|
|
|
|
Use the configured addresspool to allocate multicast addresses.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
address-pool: add clear method
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
address-pool: small cleanups
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/addresspool.c:
|
|
|
|
|
|
tests: add addresspool unit test
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-address-pool.h:
|
|
|
|
|
|
address-pool: add object to manage multicast addresses
|
|
|
|
|
|
Make an object that can manage a rage of multicast addresses and ports.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: set default max-threads property
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: wait for concurrent _prepare
|
|
|
|
|
|
If a prepare is busy, wait for the result.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: add lock around message handler
|
|
|
|
|
|
We don't want to dispatch messages while we are still processing the result of
|
|
|
|
|
|
the state change.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add lock to protect state changes
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream-transport: add keep-alive method
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
stream-transport: add method to handle RTP/RTCP
|
|
|
|
|
|
Call new methods instead of poking into the structures directly.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
session-media: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
session: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: free old socket
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
mapping: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
auth: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: add max-thread property
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: use a threadpool for the mainloops
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: rename method
|
|
|
|
|
|
gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
|
|
|
|
|
|
don't really create the client from the socket, we use the socket for the
|
|
|
|
|
|
client.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: rework maincontext handling in clients
|
|
|
|
|
|
Make a separate method to attach a client to a MainContext.
|
|
|
|
|
|
Let the server decide in what GMainContext the client will operate and give this
|
|
|
|
|
|
context to the client in attach. Then the server can later decide to use a
|
|
|
|
|
|
separate thread for each client or just use the mainthread.
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
session: move session header code in session object
|
|
|
|
|
|
|
|
|
|
|
|
2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* COPYING:
|
|
|
|
|
|
* COPYING.LIB:
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
* tests/test-cleanup.c:
|
|
|
|
|
|
Fix FSF address
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
rtsp-server: added annotations to indicate type of ownership transfer of return values
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=680777
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* bindings/Makefile.am:
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.deps:
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.deps:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.files:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.namespace:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
bindings: remove vala bindings
|
|
|
|
|
|
They'll be reunited with the other GStreamer bindings
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=680777
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
rtsp: only create transport when needed
|
|
|
|
|
|
Only create the StreamTransport when configured.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: small cleanup
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
rtsp: refactor configuration of transport
|
|
|
|
|
|
Move the configuration of the transport to a place where it makes
|
|
|
|
|
|
more sense.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: refactor transport parsing
|
|
|
|
|
|
|
|
|
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|
2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
|
|
client: refuse to change the MTU on shared media
|
|
|
|
|
|
If we change the MTU of chared media, it changes for all clients.
|
|
|
|
|
|
We don't want to set the MTU to something large for clients that
|
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|
|
|
stream over UDP.
|
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|
|
2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
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|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
small fixes to docs and debug
|
|
|
|
|
|
|
|
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|
|
2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
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|
|
|
stream: transports must already have been removed
|
|
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|
|
2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
stream: improve join and leave of the pipeline
|
|
|
|
|
|
simplify code
|
|
|
|
|
|
Do the cleanup properly
|
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|
|
Add some docs
|
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|
|
2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
|
media: move unprepare below default implementation
|
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|
|
Makes it easier to find the default implementation
|
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|
|
2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
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|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
media: signal unprepared when we actually finish
|
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|
|
2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
|
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|
|
media: no need to unlock, unprepare does that when needed
|
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|
2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
docs: update docs
|
|
|
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|
|
|
|
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|
|
|
2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp: fix MTU setting
|
|
|
|
|
|
Fix setting of the MTU. There is no need for a vmethod.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
docs: update docs
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump version number after refactoring
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.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-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream-transport.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-stream.h:
|
|
|
|
|
|
rtsp: massive refactoring
|
|
|
|
|
|
Make GObjects from the remaining simple structures.
|
|
|
|
|
|
Remove GstRTSPSessionStream, it's not needed.
|
|
|
|
|
|
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
|
|
|
|
|
|
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
|
|
|
|
|
|
a GstRTSPStream should be transported to a client.
|
|
|
|
|
|
Rename GstRTSPMediaFactory::get_element -> create_element because that
|
|
|
|
|
|
more accurately describes what it does.
|
|
|
|
|
|
Make nice methods instead of poking in the structures.
|
|
|
|
|
|
Move some methods inside the relevant object source code.
|
|
|
|
|
|
Use GPtrArray to store objects instead of plain arrays, it is more
|
|
|
|
|
|
natural and allows us to more easily clean up.
|
|
|
|
|
|
Move the allocation of udp ports to the Stream object. The Stream object
|
|
|
|
|
|
contains the elements needed to stream the media to a client.
|
|
|
|
|
|
Improve the prepare and unprepare methods. Unprepare should now undo
|
|
|
|
|
|
everything prepare did. Improve also async unprepare when doing EOS on
|
|
|
|
|
|
shutdown. Make sure we always unprepare correctly.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: Unref server address clients connected to
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: don't ref server socket if it is NULL
|
|
|
|
|
|
Fixes test_bind_already_in_use unit test again after commit 6a497440.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=686644
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
tests: Add libgio link dependency
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
rtsp-media-mapping: rename find_media vfunc to find_factory
|
|
|
|
|
|
The virtual method and class method should have the same name
|
|
|
|
|
|
so it is correctly represented in GIR file
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=680777
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* 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-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
rtsp-server: fixed comments and GIR annotations
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=680777
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: allow binding on port 0 (binds on a random port)
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
rtsp-server: add bound-port property
|
|
|
|
|
|
bound-port can be used to retrieve the port number when the server is bound on
|
|
|
|
|
|
port 0, which binds on a random port.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
rtsp-media-factory: make ::get_element overridable by GI bindings
|
|
|
|
|
|
The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
|
|
|
|
|
|
for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
|
|
|
|
|
|
as the invoker for ::get_element(), making it overridable by GI generated
|
|
|
|
|
|
bindings.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
rtsp-media-factory-uri: don't autoplug parsers in a loop
|
|
|
|
|
|
Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
|
|
|
|
|
|
h264parse forever.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Explicitly link against gio. Fix link error on mac.
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
session: add ttl to the transport header in SETUP
|
|
|
|
|
|
See https://bugzilla.gnome.org/show_bug.cgi?id=685561
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
client: Use client transport settings for multicast if allowed.
|
|
|
|
|
|
This patch makes it possible for the client to send transport settings for
|
|
|
|
|
|
multicast (destination && ttl). Client settings must be explicitly allowed or
|
|
|
|
|
|
the server will use its own settings.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6c0b52c to 6bb6951
|
|
|
|
|
|
|
|
|
|
|
|
2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: do not destroy the rtsp watch
|
|
|
|
|
|
Don't destroy the client watch while dispatching. The rtsp watch is
|
|
|
|
|
|
automatically destroyed after the rtsp watch function closed() has
|
|
|
|
|
|
been called.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 4f962f7 to 6c0b52c
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix check for seekability
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use more GIO
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: remove obsolete includes
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
|
|
|
|
|
|
be available in "on_new_ssrc". The transports are added in
|
|
|
|
|
|
gst_rtsp_media_set_state when going to PLAYING state. However,
|
|
|
|
|
|
"on_new_ssrc" might be called before this happens.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=683304
|
|
|
|
|
|
|
|
|
|
|
|
2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
rtsp-client: add signals for rtsp requests (fixes #683287)
|
|
|
|
|
|
|
|
|
|
|
|
2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
add new-session signal to rtsp-client (fixes #683058)
|
|
|
|
|
|
|
|
|
|
|
|
2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
|
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|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 668acee to 4f962f7
|
|
|
|
|
|
|
|
|
|
|
|
2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtsp-server: fixed segfault in gst_rtsp_server_create_socket
|
|
|
|
|
|
Do not assume that *error is set in g_socket_address_enumerator_next.
|
|
|
|
|
|
Added test_bind_already_in_use unit-test.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
|
|
|
|
|
|
|
|
|
|
|
|
2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 94ccf4c to 668acee
|
|
|
|
|
|
|
|
|
|
|
|
2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
rtsp-client: make create_sdp virtual method
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
|
|
|
|
|
|
|
|
|
|
|
|
2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 98e386f to 94ccf4c
|
|
|
|
|
|
|
|
|
|
|
|
2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix docs
|
|
|
|
|
|
|
|
|
|
|
|
2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
rtsp-server: use an existing socket to establish HTTP tunnel
|
|
|
|
|
|
Make it possible to transfer a socket from an HTTP server to be used as
|
|
|
|
|
|
an RTSP over HTTP tunnel.
|
|
|
|
|
|
|
|
|
|
|
|
2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
rtsp: Handle the blocksize parameter
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
Have unit test get header from source dir, not installed dir
|
|
|
|
|
|
This makes compilation of unit tests work in a build directory other
|
|
|
|
|
|
than the source directory.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: update for gst_element_make_from_uri() changes
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* tests/Makefile.am:
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
|
|
* tests/check/gst/rtspserver.c:
|
|
|
|
|
|
rtsp: add unit test
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: don't collect media stats when going to NULL
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: don't leak transports
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: free transport on no_stream in SETUP handler
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: changed session media iteration
|
|
|
|
|
|
In client_unlink_session: now don't iterate in session->medias
|
|
|
|
|
|
list where items are removed by gst_rtsp_session_release_media.
|
|
|
|
|
|
Instead, repeatedly remove the first item.
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
|
|
|
|
|
|
GstRTSPSessionMedia is not a GObject type. When the
|
|
|
|
|
|
GstRTSPSession is freed, it will free the media.
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
factory: plug pad leak in collect_streams
|
|
|
|
|
|
In gst_rtsp_media_factory_collect_streams: unref the srcpad that
|
|
|
|
|
|
was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
|
|
|
|
|
|
will take one reference, and the other reference will otherwise
|
|
|
|
|
|
give a memory leak.
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: suppress some warnings when debug is disabled
|
|
|
|
|
|
Warnings about unused variables should be suppressed if core has the
|
|
|
|
|
|
debug system disabled.
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
docs: fix build in uninstalled setup
|
|
|
|
|
|
Include gst-plugins-base libs properly.
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server.types:
|
|
|
|
|
|
docs: include headers defining rtsp-server object types
|
|
|
|
|
|
Fixes compiler warnings during docs build.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=676824
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: Add warning flags for compiler when configuring
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 03a0e57 to 98e386f
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 1fab359 to 03a0e57
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix GSocketAddress leak in gst_rtsp_client_accept
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
|
|
|
|
|
|
|
|
|
|
|
|
2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f1b5a96 to 1fab359
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 92b7266 to f1b5a96
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From ec1c4a8 to 92b7266
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 3429ba6 to ec1c4a8
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp: fix compiler warnings
|
|
|
|
|
|
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From dc70203 to 3429ba6
|
|
|
|
|
|
|
|
|
|
|
|
2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
rtsp-server: port to new thread API
|
|
|
|
|
|
|
|
|
|
|
|
2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6db25be to dc70203
|
|
|
|
|
|
|
|
|
|
|
|
2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-server: Fix compilation and compiler warnings
|
|
|
|
|
|
|
|
|
|
|
|
2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
configure: Modernize autotools setup a bit
|
|
|
|
|
|
Also we now only create tar.bz2 and tar.xz tarballs.
|
|
|
|
|
|
|
|
|
|
|
|
2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 464fe15 to 6db25be
|
|
|
|
|
|
|
|
|
|
|
|
2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* common:
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Automatic update of common submodule
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From 7fda524 to 464fe15
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2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* configure.ac:
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* docs/libs/Makefile.am:
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* docs/version.entities.in:
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* gst-rtsp.spec.in:
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* gst/rtsp-server/Makefile.am:
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* pkgconfig/Makefile.am:
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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* pkgconfig/gstreamer-rtsp-server.pc.in:
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* tests/Makefile.am:
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rtsp-server: Update versioning
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2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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Merge remote-tracking branch 'origin/0.10'
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Conflicts:
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gst/rtsp-server/rtsp-session-pool.c
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2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/rtsp-server/rtsp-session-pool.c:
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rtsp-server: Don't use deprecated GLib API
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2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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Replace master with 0.11
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2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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Merge branch 'master' into 0.11
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2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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Merge branch 'master' into 0.11
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2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* docs/README:
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A couple minor typo fixes
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2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media.c:
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media: fix state of the appqueue
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2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media-factory-uri.c:
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factory: use videoconvert
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2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media-factory-uri.c:
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factory: change to new style caps
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2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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* gst/rtsp-server/rtsp-media-factory-uri.c:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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* gst/rtsp-server/rtsp-session-pool.c:
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rtsp-server: port to GIO
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Port to GIO
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2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* configure.ac:
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configure: fix build
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2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* docs/README:
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docs: fix for gst_rtsp_server_set_port() -> _set_service()
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https://bugzilla.gnome.org/show_bug.cgi?id=666548
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2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* configure.ac:
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* examples/Makefile.am:
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First rule of gst-rtsp-server club: don't talk about gst-phonon
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2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* configure.ac:
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* pkgconfig/Makefile.am:
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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|
* pkgconfig/gstreamer-rtsp-server.pc.in:
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pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
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For consistency with all other modules.
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2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: update for new map API
|
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|
2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* .gitignore:
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|
* bindings/Makefile.am:
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* bindings/python/Makefile.am:
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* bindings/python/arg-types.py:
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* bindings/python/codegen/Makefile.am:
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* bindings/python/codegen/__init__.py:
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|
* bindings/python/codegen/argtypes.py:
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|
* bindings/python/codegen/code-coverage.py:
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* bindings/python/codegen/codegen.py:
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* bindings/python/codegen/definitions.py:
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* bindings/python/codegen/defsparser.py:
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|
* bindings/python/codegen/docextract.py:
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|
* bindings/python/codegen/docgen.py:
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|
* bindings/python/codegen/fileprefix.override:
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|
* bindings/python/codegen/fileprefixmodule.c:
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|
* bindings/python/codegen/h2def.py:
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|
* bindings/python/codegen/mergedefs.py:
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|
* bindings/python/codegen/mkskel.py:
|
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|
* bindings/python/codegen/override.py:
|
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|
* bindings/python/codegen/reversewrapper.py:
|
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|
|
* bindings/python/codegen/scmexpr.py:
|
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|
|
* bindings/python/rtspserver-types.defs:
|
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|
|
* bindings/python/rtspserver.defs:
|
|
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|
|
* bindings/python/rtspserver.override:
|
|
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|
|
|
* bindings/python/rtspservermodule.c:
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|
|
* bindings/python/test.py:
|
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|
|
* configure.ac:
|
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|
|
python: remove pygst-based python bindings
|
|
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|
|
pygi is the future, apparently.
|
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|
|
2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
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|
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|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From c463bc0 to 7fda524
|
|
|
|
|
|
|
|
|
|
|
|
2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 2a59016 to c463bc0
|
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|
|
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|
|
|
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|
|
2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
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|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 0807187 to 2a59016
|
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|
|
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|
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|
|
2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 11f0cd5 to 0807187
|
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|
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|
|
2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-auth.c:
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|
|
example: update for new caps
|
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|
|
2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
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|
|
|
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|
|
* examples/test-video.c:
|
|
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
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|
|
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|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
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|
|
rtsp-server: port some more to 0.11
|
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|
|
Fix caps.
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|
|
Remove bufferlist stuff
|
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|
|
Update for new API.
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|
|
Add queue before appsink now that preroll-queue-len is gone.
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|
Update for request pad changes.
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|
2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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|
Merge branch 'master' into 0.11
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|
2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
|
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|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
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|
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bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
|
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|
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
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|
2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
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* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
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bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
|
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Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
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2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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|
Merge branch 'master' into 0.11
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|
2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
* gst/rtsp-server/rtsp-media.h:
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|
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|
|
media: add a seekable boolean
|
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|
|
Maintain the seekable state with a new variable instead of reusing the
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|
|
is_live variable.
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|
|
2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
|
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|
|
* gst/rtsp-server/rtsp-media.c:
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|
|
Disallow seek in live media
|
|
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|
|
2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
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|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
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|
|
2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
#ifdef statements for windows socket creation were missing
|
|
|
|
|
|
|
|
|
|
|
|
2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From a39eb83 to 11f0cd5
|
|
|
|
|
|
|
|
|
|
|
|
2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 605cd9a to a39eb83
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use method to access property
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add protocols property
|
|
|
|
|
|
Add a property to configure the allowed protocols in the media created from the
|
|
|
|
|
|
factory.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add media-configure signal
|
|
|
|
|
|
Add signal to allow the application to configure the media after it was created
|
|
|
|
|
|
from the factory.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use method to access property
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add protocols property
|
|
|
|
|
|
Add a property to configure the allowed protocols in the media created from the
|
|
|
|
|
|
factory.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add media-configure signal
|
|
|
|
|
|
Add signal to allow the application to configure the media after it was created
|
|
|
|
|
|
from the factory.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use media multicast group
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
retab some .h
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
sdp: copy and free the server ip address
|
|
|
|
|
|
Copy and free the server ip address to make memory management easier later.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: configure multicast in media
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add property for multicast group
|
|
|
|
|
|
Add a property to configure the multicast group in the media.
|
|
|
|
|
|
Based on patches from Marc Leeman and Robert Krakora.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add property for multicast group
|
|
|
|
|
|
Add a property to configure the multicast group in the media factory.
|
|
|
|
|
|
Based on patches from Marc Leeman and Robert Krakora.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: do configuration of transport in one place
|
|
|
|
|
|
Move the configuration of the transport destination address to where we also
|
|
|
|
|
|
configure the other bits.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use media multicast group
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
retab some .h
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
sdp: copy and free the server ip address
|
|
|
|
|
|
Copy and free the server ip address to make memory management easier later.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: configure multicast in media
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add property for multicast group
|
|
|
|
|
|
Add a property to configure the multicast group in the media.
|
|
|
|
|
|
Based on patches from Marc Leeman and Robert Krakora.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add property for multicast group
|
|
|
|
|
|
Add a property to configure the multicast group in the media factory.
|
|
|
|
|
|
Based on patches from Marc Leeman and Robert Krakora.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: do configuration of transport in one place
|
|
|
|
|
|
Move the configuration of the transport destination address to where we also
|
|
|
|
|
|
configure the other bits.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: destroy pipeline on client disconnect with no prior TEARDOWN.
|
|
|
|
|
|
The problem occurs when the client abruptly closes the connection without
|
|
|
|
|
|
issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
|
|
|
|
|
|
server is where the pipeline gets torn down. Since this handler is not called,
|
|
|
|
|
|
the pipeline remains and is up and running. Subsequent clients get their own
|
|
|
|
|
|
pipelines and if the do not issue TEARDOWNs then those pipelines will also
|
|
|
|
|
|
remain up and running. This is a resource leak.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
|
|
|
|
|
|
For example, it can be used to retrieve source elements like appsrc, in a more
|
|
|
|
|
|
convenient way than subclassing get_element.
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: hold on to reference while using object
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: use new api
|
|
|
|
|
|
|
|
|
|
|
|
2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: use unstable api
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix reference counting
|
|
|
|
|
|
|
|
|
|
|
|
2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
fix compiler warnings about unused variables
|
|
|
|
|
|
|
|
|
|
|
|
2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
examples: tell rtsp uri when ready
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 69b981f to 605cd9a
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: update for buffer API change
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Makefile.am: 0.10 => @GST_MAJORMINOR@
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/.gitignore:
|
|
|
|
|
|
.gitignore: 0.10 => 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Makefile.am: 0.10 => @GST_MAJORMINOR@
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 9e5bbd5 to 69b981f
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From fd35073 to 9e5bbd5
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 46dfcea to fd35073
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: port to new caps API
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
Updated Vala bindings.
|
|
|
|
|
|
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Add a signal for newly connected clients.
|
|
|
|
|
|
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
|
|
|
|
|
|
|
|
|
|
|
|
2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/python/rtspserver.override:
|
|
|
|
|
|
python: override gst_rtsp_media_mapping_add_factory to fix refcounting
|
|
|
|
|
|
|
|
|
|
|
|
2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-funnel.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-funnel.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-server: port to 0.11
|
|
|
|
|
|
|
|
|
|
|
|
2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
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|
* common:
|
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|
|
|
add common
|
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|
|
2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
Merge branch 'master' into 0.11
|
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|
|
Conflicts:
|
|
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|
|
|
common
|
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|
|
|
|
configure.ac
|
|
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|
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|
|
|
2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From c3cafe1 to 46dfcea
|
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|
|
|
|
|
|
|
|
|
|
2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/python/Makefile.am:
|
|
|
|
|
|
* bindings/python/rtspserver.defs:
|
|
|
|
|
|
python bindings: wrap GstRTSPMediaFactoryClass vfuncs
|
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|
|
2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
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|
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|
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|
|
* bindings/python/arg-types.py:
|
|
|
|
|
|
python bindings: add GstRTSPUrlParam
|
|
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|
|
|
Needed to implement MediaFactory virtual proxies
|
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|
|
2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
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|
|
|
|
|
|
|
|
|
* bindings/python/arg-types.py:
|
|
|
|
|
|
python bindings: fix returning GstRTSPUrl types
|
|
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|
|
2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
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|
|
|
|
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|
|
|
* bindings/python/arg-types.py:
|
|
|
|
|
|
python bindings: add arg type for GstRTSPUrl
|
|
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|
|
|
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|
|
2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
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|
|
|
|
|
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|
|
* bindings/python/rtspserver.defs:
|
|
|
|
|
|
python bindings: fix the definition of MediaFactory.collect_stream
|
|
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|
|
2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
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|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 1ccbe09 to c3cafe1
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 193b717 to 1ccbe09
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From b77e2bf to 193b717
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
build: Include lcov.mak to allow test coverage report generation
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From d8814b6 to b77e2bf
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6aaa286 to d8814b6
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 6aec6b9 to 6aaa286
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
autogen: wingo signed comment
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
session: use full charset for RTSP session ID
|
|
|
|
|
|
As specified in RFC 2326 section 3.4 use full valid charset to make guessing
|
|
|
|
|
|
session ID more difficult.
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=643812
|
|
|
|
|
|
|
|
|
|
|
|
2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
rtsp-server: Don't install the funnel header
|
|
|
|
|
|
|
|
|
|
|
|
2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 1de7f6a to 6aec6b9
|
|
|
|
|
|
|
|
|
|
|
|
2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: require core/base 0.10.31
|
|
|
|
|
|
Needed at least for gst_plugin_feature_rank_compare_func().
|
|
|
|
|
|
|
|
|
|
|
|
2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From f94d739 to 1de7f6a
|
|
|
|
|
|
|
|
|
|
|
|
2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: remove more unused code
|
|
|
|
|
|
|
|
|
|
|
|
2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: remove duplicate filtering
|
|
|
|
|
|
Remove the duplicate filtering code now that we have a released -good version.
|
|
|
|
|
|
Give a warning instead.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix default buffer size
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add property to configure the buffer-size
|
|
|
|
|
|
Add a property to configure the kernel UDP buffer size.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add property to configure kernel buffer sizes
|
|
|
|
|
|
Add a property to configure the kernel UDP buffer size.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: set PYGOBJECT_REQ before using it
|
|
|
|
|
|
https://bugzilla.gnome.org/show_bug.cgi?id=640641
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/Makefile.am:
|
|
|
|
|
|
docs: recursive into sub-directories on 'make upload'
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/version.entities.in:
|
|
|
|
|
|
docs: mention full version these docs are for, not just major-minor
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.8 ===
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
release 0.10.8
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp-server: clarify docs a little
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: init debug category before starting thread
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: add realm to make it more spec compliant
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: add locking
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
example: improve example docs a little
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: ensure the watch has a ref to the server
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: simpify channel function
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: simplify management of channel and source
|
|
|
|
|
|
We don't need to keep around the channel and source objects. Let the mainloop
|
|
|
|
|
|
and the source manage the source and channel respectively.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
build tests
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* tests/.gitignore:
|
|
|
|
|
|
* tests/Makefile.am:
|
|
|
|
|
|
* tests/test-cleanup.c:
|
|
|
|
|
|
tests: add tests directory and cleanup test
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
server: improve debugging in various objects
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: chain up to the parent finalize
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/python/rtspserver-types.defs:
|
|
|
|
|
|
* bindings/python/rtspserver.defs:
|
|
|
|
|
|
* bindings/python/rtspserver.override:
|
|
|
|
|
|
* bindings/python/test.py:
|
|
|
|
|
|
gst-rtsp-server: update python bindings
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use the response from the clientstate
|
|
|
|
|
|
Create the response object only once and store in the client state.
|
|
|
|
|
|
Make all methods use the state response,
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: use signal to keep track of clients
|
|
|
|
|
|
Keep track of all the clients that the server creates and remove them when they
|
|
|
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|
|
fire the 'closed' signal.
|
|
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|
|
2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: emit signal when closing
|
|
|
|
|
|
|
|
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|
|
2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-auth.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
media: enable per factory authorisations
|
|
|
|
|
|
Allow for adding a GstRTSPAuth on the factory and media level and check
|
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|
|
permissions when accessing the factory.
|
|
|
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|
|
Add hints to the auth methods for future more fine grained authorisation.
|
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|
|
|
|
Add example application for per factory authentication.
|
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|
2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.h:
|
|
|
|
|
|
rtsp-server: Pass ClientState structure arround
|
|
|
|
|
|
Pass the collected information for the ongoing request in a GstRTSPClientState
|
|
|
|
|
|
structure that we can then pass around to simplify the method arguments. This
|
|
|
|
|
|
will also be handy when we implement logging functionality.
|
|
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|
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|
|
2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: add methods to configure authorisation
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: unref auth in finalize
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: unref auth in finalize
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server.types:
|
|
|
|
|
|
docs: add more docs
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: separate create and accept
|
|
|
|
|
|
Create separate create and accept methods so that subclasses can create custom
|
|
|
|
|
|
client object.
|
|
|
|
|
|
Configure the server in the client object and prepare for keeping track of
|
|
|
|
|
|
connected clients.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: add support for setting the server.
|
|
|
|
|
|
Add support for keeping a ref to the server that started this client
|
|
|
|
|
|
connection.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
auth: fix memleak and add some docs
|
|
|
|
|
|
Fix a memleak of the basic auth token.
|
|
|
|
|
|
Add docs for the helper function
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: delegate setup of auth to the manager
|
|
|
|
|
|
Delegate the configuration of the authentication tokens to the manager object
|
|
|
|
|
|
when configured.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-auth.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
auth: add authentication object
|
|
|
|
|
|
Add an object that can check the authorization of requests.
|
|
|
|
|
|
Implement basic authentication.
|
|
|
|
|
|
Add example authentication to test-video
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: move includes back
|
|
|
|
|
|
the includes are needed for sockaddr_in.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
rtsp: move network includes where they are needed
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
rtsp-media.h: Minor corrections in comments.
|
|
|
|
|
|
Fixes #638944
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From e572c87 to f94d739
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
* docs/.gitignore:
|
|
|
|
|
|
* docs/libs/.gitignore:
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* gst/rtsp-server/.gitignore:
|
|
|
|
|
|
gitignore: updates
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
docs: We don't build ps/pdf for API reference docs
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From ccbaa85 to e572c87
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 46445ad to ccbaa85
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-funnel.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-funnel.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
funnel: rename fsfunnel to rtspfunnel
|
|
|
|
|
|
Rename the funnel to avoid conflicts with the farsight one.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/fs-funnel.c:
|
|
|
|
|
|
* gst/rtsp-server/fs-funnel.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-media: add and use fsfunnel
|
|
|
|
|
|
Add a copy of fsfunnel to the build because input-selector removed the (broken)
|
|
|
|
|
|
select-all property that we need.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
gobject-introspection: use PKG_CONFIG_PATH specified at configure time
|
|
|
|
|
|
Use PKG_CONFIG_PATH specified at configure time (if any) as well
|
|
|
|
|
|
for the g-ir-compiler, rather than just assuming the env var has
|
|
|
|
|
|
been set.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* m4/Makefile.am:
|
|
|
|
|
|
* m4/codeset.m4:
|
|
|
|
|
|
build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
gobject-introspection: fix g-i build for uninstalled setup
|
|
|
|
|
|
Requires gst-plugins-base git (> 0.10.31.2).
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
examples: add some more options and comments
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
factory-uri: use right property type
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
factory-uri: attempt to configure buffer-lists
|
|
|
|
|
|
Attempt to configure buffer lists in the payloader for improved performance.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: attempt to configure bigger UDP buffers
|
|
|
|
|
|
Attempt to configure bigger udp kernel send buffers to avoid overflowing the
|
|
|
|
|
|
send buffers with high bitrate streams.
|
|
|
|
|
|
|
|
|
|
|
|
2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use the socket length from getsockname
|
|
|
|
|
|
Use the length returned by getsockname to perform the getnameinfo call because
|
|
|
|
|
|
the size can depend on the socket type and platform.
|
|
|
|
|
|
Fixes #638723
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
docs: add uri factory to the docs
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
docs: improve docs
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
rtsp-server: add support for buffer lists
|
|
|
|
|
|
Add support for sending bufferlists received from appsink.
|
|
|
|
|
|
Fixes #635832
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
media: make method to retrieve the play range
|
|
|
|
|
|
Make a method to retrieve the playback range so that we can conditionally create
|
|
|
|
|
|
a different range for the SDP and the PLAY requests.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: add signal to notify of state changes
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
client: cleanup headers
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix typo
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
factory-uri: add support for gstpay
|
|
|
|
|
|
Add an option to prefer gstpay over decoder + raw payloader.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
factory-uri: rework the autoplugger.
|
|
|
|
|
|
Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
|
|
|
|
|
|
before payloaders.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
factory-uri: use better factory filter
|
|
|
|
|
|
Make better payloader filter based on autoplug rank and RTP use case.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 169462a to 46445ad
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: set SO_REUSEADDR before bind
|
|
|
|
|
|
Set the SO_REUSEADDR _before_ bind() to make it actually work.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: emit prepared signal when prepared
|
|
|
|
|
|
Make a 'prepared' signal and emit it when we successfully prepared the element.
|
|
|
|
|
|
This signal can be used to configure the media object after it has been prepared
|
|
|
|
|
|
for streaming.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Automatic update of common submodule
|
|
|
|
|
|
From 011bcc8 to 169462a
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
|
|
|
|
|
|
|
|
|
|
|
|
python an optional dependency
|
|
|
|
|
|
* configure.ac: Move up valgrind and g-i checks. Make the python
|
|
|
|
|
|
dependency optional, as it was before.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
Merge branch 'master' into 0.11
|
|
|
|
|
|
Conflicts:
|
|
|
|
|
|
common
|
|
|
|
|
|
configure.ac
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: update range when active clients changed
|
|
|
|
|
|
When we changed the number of active clients, update the current range
|
|
|
|
|
|
information because we want the second client connecting to a shared resource
|
|
|
|
|
|
continue from where the stream currently.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
factory-uri: add colorspace and fix pt
|
|
|
|
|
|
Rework the way we pass data to the autoplugger.
|
|
|
|
|
|
When we have raw caps, plug a converter element to make pluggin to raw
|
|
|
|
|
|
payloaders more successful.
|
|
|
|
|
|
Make sure all dynamically plugged payloaders have a unique payload types.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-uri.c:
|
|
|
|
|
|
example: add example of the uri factory
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory-uri.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
factory-uri: add a factory to stream any URI
|
|
|
|
|
|
Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
|
|
|
|
|
|
when we have one.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: ignore spurious ASYNC_DONE messages
|
|
|
|
|
|
When we are dynamically adding pads, the addition of the udpsrc elements will
|
|
|
|
|
|
trigger an ASYNC_DONE. We have to ignore this because we only want to react to
|
|
|
|
|
|
the real ASYNC_DONE when everything is prerolled.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
media-factory: make lock macro
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-server: Remove unused variable and dead assignment
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
examples: Run gst-indent
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
rtsp-server: Run gst-indent
|
|
|
|
|
|
Since it wasn't using the upstream common previously, there was no
|
|
|
|
|
|
indentation check before commiting.
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
rtsp-server: Some more doc fixups
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
Makefile: Add cruft-cleaning support
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* docs/Makefile.am:
|
|
|
|
|
|
* docs/libs/Makefile.am:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-docs.sgml:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server-sections.txt:
|
|
|
|
|
|
* docs/libs/gst-rtsp-server.types:
|
|
|
|
|
|
* docs/version.entities.in:
|
|
|
|
|
|
docs: Add gtk-doc build system
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Makefile.am: Use standard GIR make behaviour
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
autogen/configure: Bring more in sync to standard gst module behaviour
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: warn and fail when gstrtpbin is not found
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: open 0.11 branch
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitmodules:
|
|
|
|
|
|
* common:
|
|
|
|
|
|
Add common submodule
|
|
|
|
|
|
|
|
|
|
|
|
2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
|
|
|
|
* common/ChangeLog:
|
|
|
|
|
|
* common/Makefile.am:
|
|
|
|
|
|
* common/c-to-xml.py:
|
|
|
|
|
|
* common/check.mak:
|
|
|
|
|
|
* common/coverage/coverage-report-entry.pl:
|
|
|
|
|
|
* common/coverage/coverage-report.pl:
|
|
|
|
|
|
* common/coverage/coverage-report.xsl:
|
|
|
|
|
|
* common/coverage/lcov.mak:
|
|
|
|
|
|
* common/gettext.patch:
|
|
|
|
|
|
* common/glib-gen.mak:
|
|
|
|
|
|
* common/gst-autogen.sh:
|
|
|
|
|
|
* common/gst-xmlinspect.py:
|
|
|
|
|
|
* common/gst.supp:
|
|
|
|
|
|
* common/gstdoc-scangobj:
|
|
|
|
|
|
* common/gtk-doc-plugins.mak:
|
|
|
|
|
|
* common/gtk-doc.mak:
|
|
|
|
|
|
* common/m4/.gitignore:
|
|
|
|
|
|
* common/m4/Makefile.am:
|
|
|
|
|
|
* common/m4/README:
|
|
|
|
|
|
* common/m4/as-ac-expand.m4:
|
|
|
|
|
|
* common/m4/as-auto-alt.m4:
|
|
|
|
|
|
* common/m4/as-compiler-flag.m4:
|
|
|
|
|
|
* common/m4/as-compiler.m4:
|
|
|
|
|
|
* common/m4/as-docbook.m4:
|
|
|
|
|
|
* common/m4/as-libtool-tags.m4:
|
|
|
|
|
|
* common/m4/as-libtool.m4:
|
|
|
|
|
|
* common/m4/as-python.m4:
|
|
|
|
|
|
* common/m4/as-scrub-include.m4:
|
|
|
|
|
|
* common/m4/as-version.m4:
|
|
|
|
|
|
* common/m4/ax_create_stdint_h.m4:
|
|
|
|
|
|
* common/m4/check.m4:
|
|
|
|
|
|
* common/m4/glib-gettext.m4:
|
|
|
|
|
|
* common/m4/gst-arch.m4:
|
|
|
|
|
|
* common/m4/gst-args.m4:
|
|
|
|
|
|
* common/m4/gst-check.m4:
|
|
|
|
|
|
* common/m4/gst-debuginfo.m4:
|
|
|
|
|
|
* common/m4/gst-default.m4:
|
|
|
|
|
|
* common/m4/gst-doc.m4:
|
|
|
|
|
|
* common/m4/gst-error.m4:
|
|
|
|
|
|
* common/m4/gst-feature.m4:
|
|
|
|
|
|
* common/m4/gst-function.m4:
|
|
|
|
|
|
* common/m4/gst-gettext.m4:
|
|
|
|
|
|
* common/m4/gst-glib2.m4:
|
|
|
|
|
|
* common/m4/gst-libxml2.m4:
|
|
|
|
|
|
* common/m4/gst-plugindir.m4:
|
|
|
|
|
|
* common/m4/gst-valgrind.m4:
|
|
|
|
|
|
* common/m4/gtk-doc.m4:
|
|
|
|
|
|
* common/m4/introspection.m4:
|
|
|
|
|
|
* common/m4/pkg.m4:
|
|
|
|
|
|
* common/mangle-tmpl.py:
|
|
|
|
|
|
* common/plugins.xsl:
|
|
|
|
|
|
* common/po.mak:
|
|
|
|
|
|
* common/release.mak:
|
|
|
|
|
|
* common/scangobj-merge.py:
|
|
|
|
|
|
* common/upload.mak:
|
|
|
|
|
|
common: Remove static version
|
|
|
|
|
|
|
|
|
|
|
|
2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
|
|
|
|
|
|
|
|
|
|
|
|
* common/m4/introspection.m4:
|
|
|
|
|
|
Update introspection.m4 to match usage
|
|
|
|
|
|
|
|
|
|
|
|
2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* README:
|
|
|
|
|
|
README: update
|
|
|
|
|
|
Remove old stuff from the README
|
|
|
|
|
|
|
|
|
|
|
|
2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.7 ===
|
|
|
|
|
|
|
|
|
|
|
|
2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
release 0.10.7
|
|
|
|
|
|
|
|
|
|
|
|
2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
test-ogg: remove parsers
|
|
|
|
|
|
Remove the parsers, they are not needed anymore as oggdemux now outputs normal
|
|
|
|
|
|
buffers with timestamps. Using the parsers also seems to break things.
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
Updated Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* common/m4/introspection.m4:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Added initial gobject-introspection support
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: don't use host for shared hash key
|
|
|
|
|
|
When we generate the key to share made between connections, don't include the
|
|
|
|
|
|
host used to connect so that we can share media even if between clients that
|
|
|
|
|
|
connected with localhost and ones with the ip address.
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
build: fix distcheck
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
Update Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Fix configure checks and installation location for Vala bindings
|
|
|
|
|
|
Fixes bug #628676.
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.6 ===
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: release 0.10.6
|
|
|
|
|
|
|
|
|
|
|
|
2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: help the compiler a little
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
media: cleanup media transport before freeing
|
|
|
|
|
|
Cleanup the media transport data before freeing. In particular, remove the qdata
|
|
|
|
|
|
from the rtpsource object.
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
media-factory: add eos-shutdown property
|
|
|
|
|
|
Add an eos-shutdown property that will send an EOS to the pipeline before
|
|
|
|
|
|
shutting it down. This allows for nice cleanup in case of a muxer.
|
|
|
|
|
|
Fixes #625597
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: use multiudpsink send-duplicates when we can
|
|
|
|
|
|
If we have a new enough multiudpsink with the send-duplicates property, use this
|
|
|
|
|
|
instead of doing our own filtering. Our custom filtering code should eventually
|
|
|
|
|
|
be removed when we can depend on a released -good.
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't leak destinations
|
|
|
|
|
|
Refactor and cleanup the destinations array when the stream is destroyed.
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: don't add udp addresses multiple times
|
|
|
|
|
|
Keep track of the udp addresses we added to udpsink and never add the same udp
|
|
|
|
|
|
destination twice. This avoids duplicate packets when using multicast.
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: disable use of SO_LINGER
|
|
|
|
|
|
SO_LINGER cause the client to fail to receive a TEARDOWN message because the
|
|
|
|
|
|
server close()s the connection.
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: use 5 second linger period in SO_LINGER
|
|
|
|
|
|
Wait 5 seconds before clearing the send buffers and reseting the connection with
|
|
|
|
|
|
the client when we do a close. This should be enough time to get the message to
|
|
|
|
|
|
the client.
|
|
|
|
|
|
See #622757
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: use SO_LINGER
|
|
|
|
|
|
SO_LINGER on the socket will make sure that any pending data on the socket is
|
|
|
|
|
|
flushed ASAP and that the socket connection is reset. This makes sure that the
|
|
|
|
|
|
socket can be reused immediately.
|
|
|
|
|
|
Fixes 622757
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
README: add blurb about shared media factories
|
|
|
|
|
|
|
|
|
|
|
|
2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Add stdlib.h for atoi()
|
|
|
|
|
|
|
|
|
|
|
|
2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/python/Makefile.am:
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
build: distcheck fixes
|
|
|
|
|
|
Fix 'make distcheck', somewhat (it still fails because it tries to
|
|
|
|
|
|
install files into /usr/share/vala/vapi/ irrespective of the
|
|
|
|
|
|
configured prefix).
|
|
|
|
|
|
|
|
|
|
|
|
2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump core/base requirements to released version
|
|
|
|
|
|
Makes things less confusing for people.
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: fail if GStreamer core/base requirements are not met
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: improve client cleanups
|
|
|
|
|
|
Make sure the session does not timeout when using TCP. We need to do this
|
|
|
|
|
|
because quicktime player does not send RTCP for some reason in tunneled
|
|
|
|
|
|
mode.
|
|
|
|
|
|
Refactor some cleanup code.
|
|
|
|
|
|
Fixes #612915
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
session: add support for prevent session timeouts
|
|
|
|
|
|
Add an atomix counter to prevent session timeouts when we are, for example,
|
|
|
|
|
|
streaming over TCP.
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: fix unlink on session timeouts
|
|
|
|
|
|
When our session times out, make sure we unlink all streams in this
|
|
|
|
|
|
session.
|
|
|
|
|
|
Remove the tunnelid when closing the connection.
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
session: small cleanups
|
|
|
|
|
|
|
|
|
|
|
|
2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: handle lost_tunnel callbacks
|
|
|
|
|
|
Handle lost_tunnel callbacks and use it to store the tunnelid back into the
|
|
|
|
|
|
hashtable so that we can reuse it for when the client reopens the POST
|
|
|
|
|
|
socket.
|
|
|
|
|
|
Close the connection after a TEARDOWN.
|
|
|
|
|
|
Make sure or watchid is cleared when the watch is removed.
|
|
|
|
|
|
Fixes #612915
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
rtsp-server: add more support for multicast
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: allow configuration of allowed lower transport
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
rtsp: keep track of server ip and ipv6
|
|
|
|
|
|
Keep track of how the client connected to the server and setup the udp ports
|
|
|
|
|
|
with the same protocol.
|
|
|
|
|
|
Copy the server ip address in the SDP so that clients can send RTCP back to
|
|
|
|
|
|
us.
|
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|
|
2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
session: indent
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
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|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use right size for malloc
|
|
|
|
|
|
|
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|
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|
|
2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
server: comment ipv6 server listening address
|
|
|
|
|
|
|
|
|
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|
|
2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
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|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: allow for ipv6 sockets
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
server: rework server part
|
|
|
|
|
|
Allow setting a bind address, make sure we can deal with ipv6.
|
|
|
|
|
|
Remove the port property and change with the service property.
|
|
|
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|
|
|
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|
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|
|
2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: update comments a little
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: make content-base better
|
|
|
|
|
|
Use the URI formatting functions to make a content-base. Also make sure that
|
|
|
|
|
|
there is a trailing / at the end.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: guard against invalid paths
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
test: catch server bind errors
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtspmedia: emit "unprepared" if _prepare fails.
|
|
|
|
|
|
Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
|
|
|
|
|
|
media object is removed from its factory's cache.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: collect media position when seek completes
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: call unlink_streams in client finalize
|
|
|
|
|
|
Fixes #599027
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: limit the time to wait to something huge
|
|
|
|
|
|
Avoid waiting forever but limit the timeout to 20 seconds.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: reindent and check for prepared status
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
media: avoid doing _get_state() for state changes
|
|
|
|
|
|
When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
|
|
|
|
|
|
until the media is prerolled or in error. This avoids doing a blocking call of
|
|
|
|
|
|
gst_element_get_state() that can cause lockups when there is an error.
|
|
|
|
|
|
Fixes #611899
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: reindent
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
media-factory: better error handling
|
|
|
|
|
|
Improve the error handling a bit.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: rework transport parsing
|
|
|
|
|
|
Rework the transport parsing code so that we can ignore transports we don't
|
|
|
|
|
|
support instead of just picking the first one we can parse.
|
|
|
|
|
|
Configure a (for now hardcoded) destination for multicast transports.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: set multicast sink parameters
|
|
|
|
|
|
Disable loop and automatic multicast join on the udpsink elements.
|
|
|
|
|
|
Add some more debug info.
|
|
|
|
|
|
Reset some state variables in the right place.
|
|
|
|
|
|
Use the right port numbers for multicast.
|
|
|
|
|
|
|
|
|
|
|
|
2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
session: handle transport setup correctly
|
|
|
|
|
|
Handle UDP, MCAST and TCP transport negotiation more correctly.
|
|
|
|
|
|
Store the server session SSRC in the transport.
|
|
|
|
|
|
|
|
|
|
|
|
2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp-client: implement error_full
|
|
|
|
|
|
Implement error_full to avoid some segfaults when the rtspconnection calls it.
|
|
|
|
|
|
See #608245
|
|
|
|
|
|
|
|
|
|
|
|
2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
docs: update docs and comments
|
|
|
|
|
|
|
|
|
|
|
|
2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: make server work better when behind a proxy
|
|
|
|
|
|
|
|
|
|
|
|
2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
|
|
|
|
|
|
|
|
|
|
|
|
2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
Use GStreamer's debugging subsystem
|
|
|
|
|
|
|
|
|
|
|
|
2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
|
|
|
|
|
|
|
|
|
|
|
|
2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.5 ===
|
|
|
|
|
|
|
|
|
|
|
|
2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
release 0.10.5
|
|
|
|
|
|
|
|
|
|
|
|
2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump required versions
|
|
|
|
|
|
|
|
|
|
|
|
2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: call weak-unref on client->sessions from finalize
|
|
|
|
|
|
Fixes bug #596305
|
|
|
|
|
|
|
|
|
|
|
|
2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: Fixed crasher where caps got unref'ed too often
|
|
|
|
|
|
|
|
|
|
|
|
2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* pkgconfig/.gitignore:
|
|
|
|
|
|
* pkgconfig/Makefile.am:
|
|
|
|
|
|
* pkgconfig/gst-rtsp-server-uninstalled.pc.in:
|
|
|
|
|
|
Added pkg-config file to use gst-rtsp-server uninstalled
|
|
|
|
|
|
|
|
|
|
|
|
2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: add some docs
|
|
|
|
|
|
|
|
|
|
|
|
2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
rtsp: Use gst_rtsp_watch_send_message().
|
|
|
|
|
|
Use gst_rtsp_watch_send_message() since the old API which used
|
|
|
|
|
|
gst_rtsp_watch_queue_message() has been deprecated.
|
|
|
|
|
|
|
|
|
|
|
|
2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.4 ===
|
|
|
|
|
|
|
|
|
|
|
|
2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Release 0.10.4
|
|
|
|
|
|
|
|
|
|
|
|
2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
rtsp: allocate channels in TCP mode
|
|
|
|
|
|
When the client does not provide us with channels in TCP mode, allocate channels
|
|
|
|
|
|
ourselves.
|
|
|
|
|
|
|
|
|
|
|
|
2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: don't crash when tunnelid is missing
|
|
|
|
|
|
When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
|
|
|
|
|
|
don't crash but return an error response to the client.
|
|
|
|
|
|
Fixes #589489
|
|
|
|
|
|
|
|
|
|
|
|
2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
bindings: update vala bindings with new method
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
sessionpool: add function to filter sessions
|
|
|
|
|
|
Add generic function to retrieve/remove sessions.
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
configure: bump core/base requirements to release
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix indentation
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
|
Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
|
|
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|
|
2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
set state and remove elements of media in for loop
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
Added gst_rtsp_media_remove_elements function to Vala bindings
|
|
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|
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|
|
2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Added gst_rtsp_media_remove_elements function
|
|
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|
|
2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
|
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|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Don't use name for gstrtpbin so we can add multiple instances to the pipeline
|
|
|
|
|
|
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|
|
2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
Updated Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Added vmethod unprepare to GstRTSPMedia
|
|
|
|
|
|
The default implementation sets the state of the pipeline to GST_STATE_NULL
|
|
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|
|
2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
Made collect_streams function public
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Added vmethod create_pipeline to GstRTSPMediaFactory
|
|
|
|
|
|
The pipeline is created in this method and the GstRTSPMedia's element is added to it
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: use g_source_destroy()
|
|
|
|
|
|
We need to use g_source_destroy() because we might have added the source to a
|
|
|
|
|
|
different main context than the default one.
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-params.h:
|
|
|
|
|
|
rtsp: prepare for handling GET/SET_PARAMETER
|
|
|
|
|
|
Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
|
|
|
|
|
|
is a body now.
|
|
|
|
|
|
Fix return codes of handlers.
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't leak session pads
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: clean up the messages a bit
|
|
|
|
|
|
|
|
|
|
|
|
2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: warn and skip streams without media
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
vala: Fixed typo in header file of RTSPMediaStream
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: fix message
|
|
|
|
|
|
Fix a debug message
|
|
|
|
|
|
Make dumping RTCP stats configurable
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: be less verbose and leak less
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: don't leak the destination address
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
rtsp: use RTCP to keep the session alive
|
|
|
|
|
|
Use the RTCP rtcp-from stats field to find the associated session and use this
|
|
|
|
|
|
to keep the session alive.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
session: add 5sec to the real session timeout
|
|
|
|
|
|
Allow the session to live 5sec longer before really timing out. This should give
|
|
|
|
|
|
clients some extra time to keep the session active.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: replay OK to GET/SET_PARAMETER
|
|
|
|
|
|
Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
|
|
|
|
|
|
so that we return OK for those requests.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: keep track of active transports
|
|
|
|
|
|
Keep track of which transport is active to avoid closing the connection too
|
|
|
|
|
|
soon.
|
|
|
|
|
|
Remove the destination transport also when going to NULL.
|
|
|
|
|
|
Print some stats about the SDES and other RTCP messages we receive from the
|
|
|
|
|
|
clients.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-sdp.c:
|
|
|
|
|
|
example: add SDP relay example
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: also count active TCP connections
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
rtsp: add support for dynamic elements
|
|
|
|
|
|
Add support for dynamic elements.
|
|
|
|
|
|
Don't set live pipelines back to paused.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
sdp: don't add encoding name when absent in caps
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: warn when we can't do RTP-Info
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
factory: factor out the stream construction
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
client: only add RTP-Info when we have the info
|
|
|
|
|
|
Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
|
|
|
|
|
|
depayloader.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.3 ===
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
release: 0.10.3
|
|
|
|
|
|
- Fixes a bug where it put the wrong verion in pkgconfig
|
|
|
|
|
|
- Link RTP and RTCP sources
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
media: link the RTP udpsrc to the session manager
|
|
|
|
|
|
Link the RTP udpsrc and the appsrc to the session manager so that they don't
|
|
|
|
|
|
shut down when the client sends a packet to open firewalls.
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* pkgconfig/gst-rtsp-server.pc.in:
|
|
|
|
|
|
Don't use hard-coded version number in pkg-config file
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
back to development
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.2 ===
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
release 0.10.2
|
|
|
|
|
|
|
|
|
|
|
|
2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
* common/m4/.gitignore:
|
|
|
|
|
|
* examples/.gitignore:
|
|
|
|
|
|
* pkgconfig/.gitignore:
|
|
|
|
|
|
add some .gitignore files
|
|
|
|
|
|
|
|
|
|
|
|
2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: seek to key frames
|
|
|
|
|
|
|
|
|
|
|
|
2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
media: emit the unprepared signal by id
|
|
|
|
|
|
Emit the unprepared signal by id instead of name and set the media as
|
|
|
|
|
|
reused.
|
|
|
|
|
|
|
|
|
|
|
|
2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
|
|
|
|
|
|
|
|
|
|
|
|
2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
Added finalize function to GstRTPSPServer to unref session pool and media mapping
|
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|
2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
|
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|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
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|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
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|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
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|
|
Updated vala bindings
|
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|
|
2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
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* gst/rtsp-server/Makefile.am:
|
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|
|
|
* gst/rtsp-server/rtsp-client.c:
|
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* gst/rtsp-server/rtsp-media.c:
|
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|
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server: use appsink and appsrc with the API
|
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|
|
Use the appsink/appsrc API instead of the signals for higher
|
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|
|
performance.
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2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
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* examples/test-ogg.c:
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|
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tests: set the payload type correctly
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|
2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media-factory.c:
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factory: connect to the unprepare signal
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|
Connect to the unprepare signal for non-reusable media so that we can remove
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them from the cache.
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2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
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* gst/rtsp-server/rtsp-media.c:
|
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* gst/rtsp-server/rtsp-media.h:
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|
media: add signal to notify of unprepare
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2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtsp-server/rtsp-media.c:
|
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* gst/rtsp-server/rtsp-media.h:
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|
|
media: more work on making the media shared
|
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|
|
Add a reusable flag to medias, indicating that they can be reused after a state
|
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change to NULL.
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|
|
Small cleanups.
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|
2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* examples/test-readme.c:
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|
|
examples: mark the example as shared for testing
|
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|
2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
* gst/rtsp-server/rtsp-media.h:
|
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|
|
client: support shared media
|
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|
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|
|
Always perform the state actions even if the target state of the pipeline is
|
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|
|
|
already correct, we still want to add/remove the transports when we are dealing
|
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|
|
with shared media.
|
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|
|
Keep a counter of the number of active transports for a media so that we can use
|
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|
|
this to perform a state change when needed.
|
|
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|
|
Perform a state change of the pipeline only when the first transport was added
|
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|
|
or when there are no active transports.
|
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|
2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
client: fix refcounting crasher
|
|
|
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|
|
Don't need to remove the weak refs in the finalize methods, they are already
|
|
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|
|
removed in the dispose.
|
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|
|
Don't register the callback with a DestroyNofity.
|
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|
2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
|
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|
|
Fix rtsp client refcount management in TCP mode.
|
|
|
|
|
|
Don't unref a client ref we never had. Fixes an unref
|
|
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|
|
|
of an already-free client object after a client
|
|
|
|
|
|
teardown request for me.
|
|
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|
|
2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-session.c:
|
|
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|
|
|
docs: fix typo in API docs
|
|
|
|
|
|
|
|
|
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|
|
2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
More seeking fixes.
|
|
|
|
|
|
Keep the udp sources in playing even if we go to paused. unlock the sources when
|
|
|
|
|
|
we shut down.
|
|
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|
|
Add some more debug info.
|
|
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|
|
Only seek when we need to.
|
|
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|
|
Keep track of the position when we go to paused.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add beginnings of seeking.
|
|
|
|
|
|
Parse the Range header and perform a seek on the pipeline for the requested
|
|
|
|
|
|
position. It's disabled currently until I figure out what's going wrong.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
allow pause requests for now.
|
|
|
|
|
|
--
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Remove weak ref on the session in teardown
|
|
|
|
|
|
We need to remove our weakref from the session when we do a teardown because
|
|
|
|
|
|
else we close the TCP connection prematurely.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
Do some more session cleanup
|
|
|
|
|
|
Make session timeout kill the TCP connection that currently watches the
|
|
|
|
|
|
session.
|
|
|
|
|
|
Remove the client timeout property.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Add TCP transports
|
|
|
|
|
|
Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
|
|
|
|
|
|
connection.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-launch.c:
|
|
|
|
|
|
Add example server that takes launch lines
|
|
|
|
|
|
Add an example server that streams any -launch line.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add support for live streams
|
|
|
|
|
|
Add support for live streams and ranges
|
|
|
|
|
|
Start on handling TCP data transfer.
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Free the pipeline before other things
|
|
|
|
|
|
---
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Only free the pending tunnel if there is one
|
|
|
|
|
|
--
|
|
|
|
|
|
|
|
|
|
|
|
2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
rtsp-server: Add support for tunneling
|
|
|
|
|
|
Add support for tunneling over HTTP.
|
|
|
|
|
|
Use new connection methods to retrieve the url.
|
|
|
|
|
|
Dispatch messages based on the message type instead of blindly
|
|
|
|
|
|
assuming it's always a request.
|
|
|
|
|
|
Keep track of the watch id so that we can remove it later.
|
|
|
|
|
|
Set the media pipeline to NULL before unreffing the pipeline.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Fix for channel -> watch rename in gstreamer
|
|
|
|
|
|
Rename the RTSPChannel to RTSPWatch and remove an unused variable.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Use ASYNC RTSP io
|
|
|
|
|
|
Use the async RTSP channels instead of spawning a new thread for each client.
|
|
|
|
|
|
If a sessionid is specified in a request, fail if we don't have the session.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
Add better debug info
|
|
|
|
|
|
Add some better debug info.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
Time out sessions
|
|
|
|
|
|
Add support for session timeouts in the example.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
Pass GTimeVal around for performance reasons
|
|
|
|
|
|
Get the current time only once and pass it around so that sessions don't have to
|
|
|
|
|
|
get the current time anymore.
|
|
|
|
|
|
Add experimental support for a GSource that dispatches when the session needs to
|
|
|
|
|
|
be cleaned up.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Add better support for session timeouts
|
|
|
|
|
|
Add a method to request the number of milliseconds when a session will timeout.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add suport for RTP manager monitoring
|
|
|
|
|
|
Add the first stage in monitoring the rtp manager.
|
|
|
|
|
|
Make sure we don't update the state to something we don't want.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Add support for session keepalive
|
|
|
|
|
|
Get and update the session timeout for all requests. get the session as early as
|
|
|
|
|
|
possible.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Handle media bus messages
|
|
|
|
|
|
Handle media bus messages in a custom mainloop and dispatch them to the
|
|
|
|
|
|
RTSPMedia objects. Let the default implementation handle some common messages.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
Some more session timeout handling
|
|
|
|
|
|
Move the session header setting code to a central place so that we always add
|
|
|
|
|
|
the timeout parameter too.
|
|
|
|
|
|
Handle timeouts by running the session cleanup code.
|
|
|
|
|
|
Stop media before cleaning up.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Add timeout property
|
|
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|
|
Add a timeout property ot the client and make the other properties into GObject
|
|
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|
|
properties.
|
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|
|
2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
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|
|
Use getters and setters in property code
|
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|
|
Use the getters and setters for the timeout property instead of locking
|
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|
|
|
ourselves.
|
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|
2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
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|
|
2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
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|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
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|
|
|
* gst/rtsp-server/rtsp-session.c:
|
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|
* gst/rtsp-server/rtsp-session.h:
|
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|
|
Add more timeout stuff
|
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|
|
Add method to check if a session is expired.
|
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Add method to perform cleanup on a session pool.
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|
|
2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
* gst/rtsp-server/rtsp-session-pool.c:
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|
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* gst/rtsp-server/rtsp-session-pool.h:
|
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|
|
* gst/rtsp-server/rtsp-session.c:
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* gst/rtsp-server/rtsp-session.h:
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|
|
Add beginnings of session timeouts and limits
|
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|
|
Add the timeout value to the Session header for unusual timeout values.
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|
|
Allow us to configure a limit to the amount of active sessions in a pool. Set a
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|
|
limit on the amount of retry we do after a sessionid collision.
|
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|
|
Add properties to the sessionid and the timeout of a session. Keep track of
|
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|
|
creation time and last access time for sessions.
|
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|
2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
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|
|
* gst/rtsp-server/rtsp-client.c:
|
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|
|
* gst/rtsp-server/rtsp-media.c:
|
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|
|
* gst/rtsp-server/rtsp-media.h:
|
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|
|
* gst/rtsp-server/rtsp-sdp.c:
|
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|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
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|
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|
|
* gst/rtsp-server/rtsp-session.h:
|
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|
|
|
|
Cleanup of sessions and more
|
|
|
|
|
|
Fix the refcounting of media and sessions in the client. Properly clean up the
|
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|
|
|
|
session data when the client performs a teardown.
|
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|
|
Add Server header to responses.
|
|
|
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|
|
Allow for multiple uri setups in one session.
|
|
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|
|
Add Range header to the PLAY response and add the range attribute to the SDP
|
|
|
|
|
|
message.
|
|
|
|
|
|
Fix the session pool remove method, it used the wrong key in the hashtable. Also
|
|
|
|
|
|
give the ownership of the sessionid to the session object.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Rename a variable
|
|
|
|
|
|
Rename the 'server_port' variable to simply 'port'.
|
|
|
|
|
|
|
|
|
|
|
|
2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Rework the way we handle transports for streams
|
|
|
|
|
|
Make the media accept an array of transports for the streams that we have
|
|
|
|
|
|
configured for the play/pause requests.
|
|
|
|
|
|
Implement server states for a client and its media.
|
|
|
|
|
|
Require 0.10.22.1 (git HEAD) of gstreamer.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
Drop const from functions dealing with urls
|
|
|
|
|
|
Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
|
|
|
|
|
|
have the right const in them.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.c:
|
|
|
|
|
|
Fix various leaks
|
|
|
|
|
|
Fix some leaks.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
More cleanups
|
|
|
|
|
|
Don't keep a reference to the GstRTSPMedia in the stream.
|
|
|
|
|
|
Free more things when freeing the GstRTSPMedia.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* 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-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
More docs and small cleanups
|
|
|
|
|
|
Add some more docs and update the README
|
|
|
|
|
|
Cleanup some method names.
|
|
|
|
|
|
Remove an unneeded idx field in the GstRTSPMediaStream
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* docs/README:
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-readme.c:
|
|
|
|
|
|
Add a README and more example code
|
|
|
|
|
|
Add a README file that contains a small introduction on how to use the server
|
|
|
|
|
|
along with the example code explained in the readme.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
Fix some leaks and change default port
|
|
|
|
|
|
Fix some memory leaks by setting the udpsrc elements to the unlocked state after
|
|
|
|
|
|
we finished the initial preroll. If we keep them locked, setting the pipeline to
|
|
|
|
|
|
NULL will not stop and clean up the sources correctly.
|
|
|
|
|
|
Change the default RTSP port to 8554 aka the official alternative RTSP port.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Cleanups to the session object
|
|
|
|
|
|
Remove some unneeded variables in the session state of a stream such as the
|
|
|
|
|
|
owner media and the server transport.
|
|
|
|
|
|
Get the configuration of a media stream in a session based on the media_stream
|
|
|
|
|
|
in the original object instead of our cached index.
|
|
|
|
|
|
Free more data in the finalize method.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Cleanups and reuse media from DESCRIBE
|
|
|
|
|
|
Handle thread create errors.
|
|
|
|
|
|
Rename some internal methods to better match what they actually do.
|
|
|
|
|
|
Handle misconfiguration of session_pool and media_mapping gracefully.
|
|
|
|
|
|
Cache the DESCRIBE media and uri in the client connection and reuse them when
|
|
|
|
|
|
we receive a SETUP request in the same connection for the same uri.
|
|
|
|
|
|
Cleanup the client connection object.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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:
|
|
|
|
|
|
Add shared properties to media and factory
|
|
|
|
|
|
Add the shared property to media.
|
|
|
|
|
|
Implement some simple caching in the factory depending on if the media is shared
|
|
|
|
|
|
or not.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Add a little comment
|
|
|
|
|
|
Add some comment about the content-base header.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* examples/test-mp4.c:
|
|
|
|
|
|
* examples/test-ogg.c:
|
|
|
|
|
|
* examples/test-video.c:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* 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-sdp.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-sdp.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Reorganize things, prepare for media sharing
|
|
|
|
|
|
Added various other test server examples
|
|
|
|
|
|
Move the SDP message generation to a separate helper.
|
|
|
|
|
|
Refactor common code for finding the session.
|
|
|
|
|
|
Add content-base for realplayer compatibility
|
|
|
|
|
|
Clean up request uris before processing for better vlc compatibility.
|
|
|
|
|
|
Move prerolling and pipeline construction to the RTSPMedia object.
|
|
|
|
|
|
Use multiudpsink for future pipeline reuse.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Back to development
|
|
|
|
|
|
Back to 0.10.1.1
|
|
|
|
|
|
|
|
|
|
|
|
=== release 0.10.1 ===
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Make 0.10.1 release
|
|
|
|
|
|
Release 0.10.1
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
Fix make dist
|
|
|
|
|
|
Add more directories and files to the dist.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/python/Makefile.am:
|
|
|
|
|
|
* bindings/python/rtspserver.override:
|
|
|
|
|
|
Fixed compile error of python bindings
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
Marked values as nullable accordingly
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.excludes:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
Updated Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
Cleanups and doc updates
|
|
|
|
|
|
Add some more documentation and do some minor cleanups here and there.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* 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-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
More improvements
|
|
|
|
|
|
Rename GstRTSPMediaBin to GstRTSPMedia
|
|
|
|
|
|
Parse the request url into a GstRTSPUri object and pass this object to the
|
|
|
|
|
|
various handlers and methods that require the uri.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/main.c:
|
|
|
|
|
|
Update example
|
|
|
|
|
|
Add some more docs and remove some old code from the example.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Handle state change failures better
|
|
|
|
|
|
Handle state change failures better when changing the state of the pipeline to
|
|
|
|
|
|
determine the SDP.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
Make element creation more extendible
|
|
|
|
|
|
Add get_element vmethod to the default MediaFactory so that subclasses can just
|
|
|
|
|
|
override that method and still use the default logic for making a MediaBin from
|
|
|
|
|
|
that.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/main.c:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-mapping.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Make the server handle arbitrary pipelines
|
|
|
|
|
|
Make GstMediaFactory an object that can instantiate GstMediaBin objects.
|
|
|
|
|
|
The GstMediaBin object has a handle to a bin with elements and to a list of
|
|
|
|
|
|
GstMediaStream objects that this bin produces.
|
|
|
|
|
|
Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
|
|
|
|
|
|
with methods to register and remove those mappings.
|
|
|
|
|
|
Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
|
|
|
|
|
|
used by the server instance.
|
|
|
|
|
|
Modify the example application so that it shows how to create custom pipelines
|
|
|
|
|
|
attached to a specific mount point.
|
|
|
|
|
|
Various misc cleanps.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Allow setting a custom media factory for a server
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Allow setting a custom media factory for a client.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Add Makefile entry for the media factory
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media-factory.h:
|
|
|
|
|
|
Add media factory to map urls to media pipeline objects.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-media.h:
|
|
|
|
|
|
Add comments. Remove unused field
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.h:
|
|
|
|
|
|
Allow custom session pools to override the session id allocation algorithms Add some comments.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.h:
|
|
|
|
|
|
Add some comments.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Move the connection code in one place Add some comments
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Make vmethod to create and accept new clients. Add some docs.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.h:
|
|
|
|
|
|
Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.h:
|
|
|
|
|
|
Name the parameters more appropriately.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session-pool.c:
|
|
|
|
|
|
Do some more cleanup of the session pool.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/rtsp-client.c:
|
|
|
|
|
|
Check if return value of gst_rtsp_session_get_media is not NULL
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Install rtsp-session and rtsp-session-pool headers
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* .gitignore:
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* acinclude.m4:
|
|
|
|
|
|
* bindings/python/Makefile.am:
|
|
|
|
|
|
* bindings/python/arg-types.py:
|
|
|
|
|
|
* bindings/python/codegen/Makefile.am:
|
|
|
|
|
|
* bindings/python/codegen/__init__.py:
|
|
|
|
|
|
* bindings/python/codegen/argtypes.py:
|
|
|
|
|
|
* bindings/python/codegen/code-coverage.py:
|
|
|
|
|
|
* bindings/python/codegen/codegen.py:
|
|
|
|
|
|
* bindings/python/codegen/definitions.py:
|
|
|
|
|
|
* bindings/python/codegen/defsparser.py:
|
|
|
|
|
|
* bindings/python/codegen/docextract.py:
|
|
|
|
|
|
* bindings/python/codegen/docgen.py:
|
|
|
|
|
|
* bindings/python/codegen/fileprefix.override:
|
|
|
|
|
|
* bindings/python/codegen/fileprefixmodule.c:
|
|
|
|
|
|
* bindings/python/codegen/h2def.py:
|
|
|
|
|
|
* bindings/python/codegen/mergedefs.py:
|
|
|
|
|
|
* bindings/python/codegen/mkskel.py:
|
|
|
|
|
|
* bindings/python/codegen/override.py:
|
|
|
|
|
|
* bindings/python/codegen/reversewrapper.py:
|
|
|
|
|
|
* bindings/python/codegen/scmexpr.py:
|
|
|
|
|
|
* bindings/python/rtspserver-types.defs:
|
|
|
|
|
|
* bindings/python/rtspserver.defs:
|
|
|
|
|
|
* bindings/python/rtspserver.override:
|
|
|
|
|
|
* bindings/python/rtspservermodule.c:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Add python bindings.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Don't go into python dir when requirements for python bindings are missing
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/Makefile.am:
|
|
|
|
|
|
* bindings/vala/Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
Install Vala bindings if vala is available
|
|
|
|
|
|
|
|
|
|
|
|
2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.deps:
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server-0.10.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.deps:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.excludes:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.files:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server-0.10.namespace:
|
|
|
|
|
|
Regenerated Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.metadata:
|
|
|
|
|
|
Fixed typo in included headers for vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
|
|
* pkgconfig/Makefile.am:
|
|
|
|
|
|
* pkgconfig/gst-rtsp-server.pc.in:
|
|
|
|
|
|
Added pkgconfig file
|
|
|
|
|
|
|
|
|
|
|
|
2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.excludes:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.metadata:
|
|
|
|
|
|
Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
|
|
|
|
|
|
|
|
|
|
|
|
2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
|
|
|
|
|
|
|
|
|
|
|
|
* bindings/vala/gst-rtsp-server.vapi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.deps:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.files:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.gi:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.metadata:
|
|
|
|
|
|
* bindings/vala/packages/gst-rtsp-server.namespace:
|
|
|
|
|
|
Added Vala bindings
|
|
|
|
|
|
|
|
|
|
|
|
2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-session.c:
|
|
|
|
|
|
Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
|
|
|
|
|
|
|
|
|
|
|
|
2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Put GStreamer version in library name
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* examples/Makefile.am:
|
|
|
|
|
|
* gst/rtsp-server/Makefile.am:
|
|
|
|
|
|
Fix some issues to pass distcheck
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* gst/rtsp-server/rtsp-server.c:
|
|
|
|
|
|
Added port property to GstRTSPServer class.
|
|
|
|
|
|
|
|
|
|
|
|
2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
|
|
* configure.ac:
|
|
|
|
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* examples/Makefile.am:
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* examples/main.c:
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* gst/Makefile.am:
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* gst/rtsp-server/Makefile.am:
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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* gst/rtsp-server/rtsp-server.c:
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* gst/rtsp-server/rtsp-server.h:
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* gst/rtsp-server/rtsp-session-pool.c:
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* gst/rtsp-server/rtsp-session-pool.h:
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* gst/rtsp-server/rtsp-session.c:
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* gst/rtsp-server/rtsp-session.h:
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* src/Makefile.am:
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Split in library and example program
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2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
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* src/rtsp-client.h:
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Removed obsolete variable
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2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
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* src/rtsp-client.c:
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* src/rtsp-client.h:
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Removed pipeline variable GstRTSPClient, because it's only used in one function
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2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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* src/rtsp-media.c:
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Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
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2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
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* src/rtsp-session.c:
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Initialize some more vars.
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2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
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* src/rtsp-session.c:
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Initialize variable to avoid compiler warning.
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2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
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* .gitignore:
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Add a reasonable generic .gitignore
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