Release 1.11.90

[platform/upstream/gstreamer.git] / ChangeLog

=== release 1.11.90 ===

2017-04-07  Sebastian Dröge <slomo@coaxion.net>

        * configure.ac:
          releasing 1.11.90

2017-03-27 18:19:33 +0100  Tim-Philipp Müller <tim@centricular.com>

        * examples/test-launch.c:
          examples: make test-launch pipeline shared by default as well

2017-02-27 19:10:44 +0200  Sebastian Dröge <sebastian@centricular.com>

        * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
          gstreamer-rtsp-server: Add both srcdir and builddir to the include path
          Just the build dir is not going to work for srcdir!=builddir.

2017-02-24 15:59:54 +0200  Sebastian Dröge <sebastian@centricular.com>

        * meson.build:
          meson: Update version

2017-02-24 15:37:49 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.11.2 ===

2017-02-24 15:10:07 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.11.2

2017-02-14 20:40:26 +0000  Tim-Philipp Müller <tim@centricular.com>

        * Makefile.am:
          meson: dist meson build files
          Ship meson build files in tarballs, so people who use tarballs
          in their builds can start playing with meson already.

2017-02-07 23:39:37 +1100  Jan Schmidt <jan@centricular.com>

        * examples/test-record.c:
          examples/test-record: Add extra line to initial printout
          Add an example line of how to deliver a stream to the
          RTSP RECORD example

2017-01-19 14:57:19 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
          If there is no Content-Length header, no body would be allocated and the
          '\0' would also not be appended to the body.

2017-01-19 14:24:07 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
          While they logically have 0 bytes length, GstRTSPConnection is appending
          a '\0' to everything making the size be 1 instead.

2017-01-13 12:39:36 +0000  Tim-Philipp Müller <tim@centricular.com>

        * meson.build:
          meson: bump version

2017-01-12 19:04:23 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-session.c:
          rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
          gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
          affected.

2017-01-12 16:32:59 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.11.1 ===

2017-01-12 16:14:46 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
        * win32/common/libgstrtspserver.def:
          Release 1.11.1

2017-01-10 08:34:50 +0100  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: corrected if-statement in _get_server_port()
          This bug was accidentally introduced while fixing a segfault
          in _get_server_port() function.
          https://bugzilla.gnome.org/show_bug.cgi?id=776345

2017-01-09 14:12:05 +0100  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-stream.c:
        * tests/check/gst/stream.c:
          rtsp-stream: fixed segmenation fault in _get_server_port()
          Calling function gst_rtsp_stream_get_server_port() results in
          segmenation fault in the RTP/RTSP/TCP case.
          Port that the server will use to receive RTCP makes only
          sense in the UDP case, however the function should handle
          the TCP case in a nicer way.
          https://bugzilla.gnome.org/show_bug.cgi?id=776345

2017-01-09 12:22:40 +0300  Aleksandr Slobodeniuk <alenuke@yandex.ru>

        * gst/rtsp-server/rtsp-media-factory.c:
          dosc: Fix a little typo
          https://bugzilla.gnome.org/show_bug.cgi?id=777037

2017-01-04 16:20:54 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

        * pkgconfig/Makefile.am:
        * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
        * pkgconfig/meson.build:
          meson: generate pkg-config -uninstalled pc files
          Generating those files is useful for users building the GStreamer stack
          using meson and having to link it to another project which is still
          using the autotools.
          https://bugzilla.gnome.org/show_bug.cgi?id=776810

2017-01-04 16:11:08 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

        * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
          pkgconfig: fix -uninstalled pc file
          pcfiledir was never defined so the paths were wrong.
          https://bugzilla.gnome.org/show_bug.cgi?id=776867

2016-12-21 13:41:50 +0100  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-stream.c:
        * tests/check/gst/rtspserver.c:
          rtsp-stream: Fixed TCP transport case
          Make sure that the appsink element is actually added to
          the bin before trying to link it with the elements in it.
          https://bugzilla.gnome.org/show_bug.cgi?id=776343

2016-12-16 17:26:04 +0000  Tim-Philipp Müller <tim@centricular.com>

        * .gitignore:
        * Makefile.am:
        * configure.ac:
        * gst-rtsp.spec.in:
          Remove generated .spec file
          Likely extremely bitrotten, and we should not ship this anyway.

2016-12-03 08:21:02 +0100  Edward Hervey <bilboed@bilboed.com>

        * common:
          Automatic update of common submodule
          From f980fd9 to 39ac2f5

2016-12-02 15:40:09 +0100  Edward Hervey <edward@centricular.com>

        * gst/rtsp-server/rtsp-media.c:
          media: Fix pt map caps
          Since decryption is handled within rtpbin, all outcoming stream
          caps will be application/x-rtp (i.e. regular rtp)
          Fixes RECORD with SRTP streams

2016-12-02 15:38:04 +0100  Edward Hervey <edward@centricular.com>

        * gst/rtsp-server/rtsp-media-factory.c:
          media-factory: Create media objects with the proper transport mode
          The function called immediately afterwards (collect_streams()) will
          need it to work properly

2016-12-02 14:36:50 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-auth.c:
          rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected

2016-12-01 18:04:34 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-media-factory.c:
          rtsp-media-factory: Don't create a pipeline for the media pipeline string
          We're going to put a pipeline into a pipeline otherwise, which is not
          exactly ideal.

2016-10-25 15:41:28 +0300  Kseniia Vasilchuk <vasilchukkseniia@gmail.com>

        * gst/rtsp-server/rtsp-media.c:
          media: Fix race condition around finish_unprepare() if called multiple time
          https://bugzilla.gnome.org/show_bug.cgi?id=755329

2016-11-30 14:06:36 +1100  Jan Schmidt <jan@centricular.com>

        * gst/rtsp-sink/gstrtspclientsink.c:
          rtspclientsink: Don't leave stale pointer after unref
          Fix a warning on shutdown - don't keep a pointer to an
          alread-unreffed object.

2016-11-26 11:24:50 +0000  Tim-Philipp Müller <tim@centricular.com>

        * .gitmodules:
          common: use https protocol for common submodule
          https://bugzilla.gnome.org/show_bug.cgi?id=775110

2016-11-21 23:29:56 +1100  Matthew Waters <matthew@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: block the output of rtpbin instead of the source pipeline
          85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
          detection of the srtp rollover counter to add to the SDP.
          Unfortunately, it was incomplete for live pipelines where the logic
          blocks the source bin before creating the SDP and thus would never have
          the necessary informaiton to create a correct SDP with srtp encryption.
          Move the pad blocks to rtpbin's output pads instead so that the
          necessary information can be created before we need the information for
          the SDP.
          https://bugzilla.gnome.org/show_bug.cgi?id=770239

2016-11-21 16:02:39 +0100  Dag Gullberg <dagg@axis.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: add IDLE timeout, before session exists
          The RTSP server will not timeout an idle RTSP connection
          (note this is different from doing timeout on a RTSP
          session).
          At least for Apache this is a problem when running RTSP over
          HTTPS since it uses one of the threads (there is a rather
          limited number) that are available for handling requests.
          https://bugzilla.gnome.org/show_bug.cgi?id=771830

2016-11-23 09:45:08 +0000  Tim-Philipp Müller <tim@centricular.com>

        * .gitignore:
          .gitignore more

2016-11-21 13:05:50 +0100  Göran Jönsson <goranjn@axis.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Set close-socket FALSE on UDP src:es
          With this RTSP server can use the sockets independent on the udpsrc
          state.
          When the udp src is finalized it will unref socket and when g_socket
          is finalized the socket will be closed.
          https://bugzilla.gnome.org/show_bug.cgi?id=765673

2016-11-18 17:47:13 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-sink/gstrtspclientsink.c:
          rtspclientsink: Move to new helper function to parse authentication responses
          https://bugzilla.gnome.org/show_bug.cgi?id=774416

2016-11-16 08:42:24 +0200  Sebastian Dröge <sebastian@centricular.com>

        * examples/Makefile.am:
        * examples/test-auth-digest.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-auth.h:
        * win32/common/libgstrtspserver.def:
          rtsp-auth: Add support for Digest authentication
          https://bugzilla.gnome.org/show_bug.cgi?id=774416

2016-11-17 09:41:53 -0800  Scott D Phillips <scott.d.phillips@intel.com>

        * Makefile.am:
        * gst/rtsp-server/meson.build:
        * meson.build:
        * tests/check/meson.build:
        * win32/MANIFEST:
        * win32/common/libgstrtspserver.def:
          Enable building with MSVC
          https://bugzilla.gnome.org/show_bug.cgi?id=774640

2016-11-18 20:23:14 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

        * meson.build:
          meson: gstreamer gst_check_dep does not exist on windows

2016-11-17 09:43:37 -0800  Scott D Phillips <scott.d.phillips@intel.com>

        * gst/rtsp-server/rtsp-client.c:
          client: update do_send_message to match type GstRTSPClientSendFunc
          This type mismatch fails building with MSVC
          https://bugzilla.gnome.org/show_bug.cgi?id=774640

2016-11-11 14:42:08 +0200  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-sdp.c:
          rtsp-sdp: Fix indentation

2016-11-10 05:16:00 +0000  Neha Arora <arora.neha@samsung.com>

        * gst/rtsp-server/rtsp-media.c:
          rtsp-media: Only signal "new-state" if the state has actually changed
          https://bugzilla.gnome.org/show_bug.cgi?id=774173

2016-08-24 11:39:13 +0200  Branko Subasic <branko@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          client: emit signal in the beginning of each rtsp request
          These signals let the application validate the requests, configure the
          media/stream in a certain way and also generate error status code in
          case of error or bad request.
          https://bugzilla.gnome.org/show_bug.cgi?id=758062

2016-11-01 18:10:35 +0000  Tim-Philipp Müller <tim@centricular.com>

        * meson.build:
          meson: update version

=== release 1.11.0 ===

2016-11-01 18:53:15 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.10.0 ===

2016-11-01 18:06:46 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.10.0

2016-10-28 18:38:01 +0100  Tim-Philipp Müller <tim@centricular.com>

        * tests/check/gst/rtspserver.c:
        * tests/check/gst/stream.c:
          tests: try to avoid using the same ports in different tests
          Causes problems with client multicast tests otherwise if
          tests are run in parallel.
          https://bugzilla.gnome.org/show_bug.cgi?id=773640

2016-10-28 17:50:59 +0100  Tim-Philipp Müller <tim@centricular.com>

        * tests/check/gst/client.c:
          tests: client: use fail_unless_equals_foo() for better failure reporting

2016-09-26 11:16:04 +0200  Göran Jönsson <goranjn@axis.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: Session filter in unwatch session
          Call session filter with filter_session_media as paramer in
          client_unwatch_session if using drop_backlog = FALSE.
          In client_unwatch_session its allowed to grow the watchs backlog.
          If using drop_backlog = FALSE and the backlog is full it will cause
          a deadlock when setting session media state to NULL
          if the backlog is not allowed to grow.
          https://bugzilla.gnome.org/show_bug.cgi?id=771983

2016-10-20 21:40:18 +0100  Tim-Philipp Müller <tim@centricular.com>

        * meson.build:
          meson: add fallbacks for gst modules
          For gst-all.

2016-09-14 17:48:39 +0300  Nikita Bobkov <NikitaDBobkov@gmail.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: Fix factory leaking in find_media() in error cases
          https://bugzilla.gnome.org/show_bug.cgi?id=771488

2016-10-06 11:47:50 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: Fix randomly missing streams from SDP with dynamic elements
          When using dynamic elements, gst_rtsp_stream_join_bin() is called from
          "pad-added" signal. In that case priv->srcpad could already have its caps,
          and they'll be sent to priv->send_src[0] pad. That means that when it
          connects "notify::caps" signal, that pad could already have received its
          caps and the signal won't be emitted anymore.
          In that case priv->caps stay to NULL and when building the SDP that stream
          gets ignored. Leading to missing video or audio when playing in client side.
          https://bugzilla.gnome.org/show_bug.cgi?id=772478

2016-09-30 11:42:08 +0100  Tim-Philipp Müller <tim@centricular.com>

        * meson.build:
          meson: update version

=== release 1.9.90 ===

2016-09-30 13:04:12 +0300  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.9.90

2016-09-17 13:17:19 +0100  Ian Jamison <ian.dev@arkver.com>

        * gst/rtsp-server/rtsp-media-factory.c:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-stream.c:
          rtsp-server: Hint that set_multicast_iface expects the name of the interface
          To prevent any possibly confusion with IPs or anything else.
          https://bugzilla.gnome.org/show_bug.cgi?id=771530

2016-09-18 09:58:55 -0400  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-media-factory.c:
        * gst/rtsp-server/rtsp-media.c:
          rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
          https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5

2016-09-14 11:31:15 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          configure: Depend on gstreamer 1.9.2.1

2016-09-10 20:52:31 +1000  Jan Schmidt <jan@centricular.com>

        * autogen.sh:
        * common:
          Automatic update of common submodule
          From b18d820 to f980fd9

2016-09-10 09:58:31 +1000  Jan Schmidt <jan@centricular.com>

        * autogen.sh:
        * common:
          Automatic update of common submodule
          From 6f2d209 to b18d820

2016-09-07 18:44:34 +0300  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Remove unused _locked() variant of a function
          It was added during refactoring.

2016-09-07 10:21:09 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: cosmetic cleanup
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-09-07 10:16:19 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: Compare IP addresses case insensitive in more places
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-09-07 10:12:18 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * common:
        * gst/rtsp-server/rtsp-stream.c:
          stream: Fix leaked joined_bin
          There is no need to keep a strong ref on it, and _leave_bin() was
          setting it to NULL before calling g_clear_object() so it was leaked.
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-09-06 19:15:23 +0300  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Compare IP address strings case insensitive
          Otherwise IPv6 addresses might fail this comparision.

2016-09-06 19:10:21 +0300  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Bind multicast sockets to ANY as before
          https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48

2016-09-05 18:31:36 +0300  Kseniia <vasilchukkseniia@gmail.com>

        * gst/rtsp-server/rtsp-session.c:
          rtsp-session: Fix segfault when doing keep-alive after removing the session
          If keep-alive happens after removing the session but before finalizing the
          stream transport, we would segfault.
          https://bugzilla.gnome.org/show_bug.cgi?id=750544

2016-09-05 18:04:50 +0300  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Always create multicast UDP elements if the protocol flag is set
          Adding them later will cause deadlocks due to
          1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
          2) adding the multicast sink
          3) waiting for it to get data to preroll again
          3) never happens because the queues after the tee are full.

2016-09-05 16:32:57 +0300  Sebastian Dröge <sebastian@centricular.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: Fix up various multicast related issues

2016-09-05 13:40:59 +0300  Sebastian Dröge <sebastian@centricular.com>

        * tests/check/gst/stream.c:
          tests: Fix compilation

2016-07-28 15:33:05 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-stream.c:
        * tests/check/gst/stream.c:
          stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
          This is basically reverting changes introduced in commit f62a9a7,
          because it was introducing various regressions:
          - It introduces a leak of udpsrc elements that got wrongly fixed by adding
          an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
          ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
          - If a mcast client connects, it creates a new socket in SETUP to try to respect
          the destination/port given by the client in the transport, and overrides the
          socket already set on the udpsink element. That means that if we already had a
          client connected, the source address on the udp packets it receives suddenly
          changes.
          - If a 2nd mcast client connects, the destination/port in its transport is
          ignored but its transport wasn't updated.
          What this patch does:
          - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
          - Always have a tee+queue when udp is enabled. This could be optimized
          again in a later patch, but is more complicated. If no unicast clients
          connects then those elements are useless, this could be also optimized
          in a later patch.
          - When mcast transport is added, it creates a new set of udpsrc/udpsink,
          seperated from those for unicast clients. Since we already support only
          one mcast address, we also create only one set of elements.
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-28 15:20:31 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: factor our plug_src function
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-21 21:46:16 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: factor out plug_sink function
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-20 23:05:09 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: small documentation clarification
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-20 15:35:44 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-14 11:10:31 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: Keep a ref on joined bin
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-20 15:11:32 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: code cleanup
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-20 23:18:23 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: small fix in error code path
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-07-20 20:09:57 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
          This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
          but keeps unit tests.
          https://bugzilla.gnome.org/show_bug.cgi?id=766612

2016-09-01 12:33:00 +0300  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.9.2 ===

2016-09-01 12:32:51 +0300  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.9.2

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

=== release 1.9.1 ===

2016-07-06 13:28:12 +0300  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.9.1

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

=== release 1.8.0 ===

2016-03-24 13:00:35 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.8.0

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

=== release 1.7.91 ===

2016-03-15 12:26:13 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.7.91

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

=== release 1.7.90 ===

2016-03-01 19:00:45 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.7.90

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

=== release 1.7.2 ===

2016-02-19 12:03:18 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.7.2

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

=== release 1.7.1 ===

2015-12-24 14:54:06 +0100  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.7.1

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

=== release 1.6.0 ===

2015-09-25 23:32:52 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.6.0

=== release 1.5.91 ===

2015-09-18 20:12:06 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.5.91

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

=== release 1.5.90 ===

2015-08-19 14:15:23 +0300  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.5.90

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

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

2015-07-20 16:37:44 -0400  Xavier Claessens <xavier.claessens@collabora.com>

        * 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

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

=== release 1.5.2 ===

2015-06-24 23:44:37 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.5.2

2015-06-18 13:12:04 +0200  Ognyan Tonchev <ognyan@axis.com>

        * 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

2015-06-16 17:50:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 6015d26 to f74b2df

2015-06-11 17:39:00 +0200  Ognyan Tonchev <ognyan@axis.com>

        * 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

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

=== release 1.5.1 ===

2015-06-07 11:20:01 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.5.1

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

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

2014-10-23 13:39:10 +0200  Linus Svensson <linussn@axis.com>

        * tests/check/gst/client.c:
          tests: Remove FIXME
          The session leak is now fixed, lets remove those FIXME comments.

2014-10-23 17:54:37 +0200  Linus Svensson <linussn@axis.com>

        * tests/check/gst/rtspserver.c:
          tests: Test to setup two sessions on one connection
          https://bugzilla.gnome.org/show_bug.cgi?id=739112

2014-10-24 12:05:27 +0200  Linus Svensson <linussn@axis.com>

        * tests/check/gst/rtspserver.c:
          tests: Test setup with tcp transport
          https://bugzilla.gnome.org/show_bug.cgi?id=739112

2014-10-24 12:04:54 +0200  Linus Svensson <linussn@axis.com>

        * gst/rtsp-server/rtsp-client.c:
          client: Configure transport after creating session media
          The default implementation of configure_client_transport() in
          rtsp-client uses the session media when it chooses channels for
          interleaved traffic.
          https://bugzilla.gnome.org/show_bug.cgi?id=739112

2014-10-23 12:54:03 +0200  Linus Svensson <linussn@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-session-media.c:
          client: Stop caching media in client when doing setup
          If the media has been managed by a session media, it should not be
          cached in the client any longer. The GstRTSPSessionMedia object is now
          responsible for unpreparing the GstRTSPMedia object using
          gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
          session media.
          https://bugzilla.gnome.org/show_bug.cgi?id=739112

2014-10-31 23:01:53 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

        * gst/rtsp-server/rtsp-stream.c:
          rtsp-stream: unref srtp decoder when leaving bin
          https://bugzilla.gnome.org/show_bug.cgi?id=739481

2014-10-29 21:01:39 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: mikey memory leaks
          https://bugzilla.gnome.org/show_bug.cgi?id=739383

2014-10-27 18:01:35 +0100  Sebastian Dröge <sebastian@centricular.com>

        * common:
          Automatic update of common submodule
          From 84d06cd to 7bb2bce

2014-10-24 17:48:04 +0100  Tim-Philipp Müller <tim@centricular.com>

        * Makefile.am:
          Parallelise 'make check-valgrind'

2014-10-21 13:04:14 +0100  Tim-Philipp Müller <tim@centricular.com>

        * common:
          Automatic update of common submodule
          From a8c8939 to 84d06cd

2014-10-21 13:00:49 +0200  Stefan Sauer <ensonic@users.sf.net>

        * common:
          Automatic update of common submodule
          From 36388a1 to a8c8939

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

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

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

=== release 1.4.0 ===

2014-07-19 17:56:31 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.4.0

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

=== release 1.3.91 ===

2014-07-11 12:19:08 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.3.91

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

=== release 1.3.90 ===

2014-06-28 11:48:29 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.3.90

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>

        * 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.

2014-06-22 19:37:31 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.3.3 ===

2014-06-22 19:36:14 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.3.3

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

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

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

2014-05-21 13:23:40 +0200  Sebastian Dröge <sebastian@centricular.com>

        * configure.ac:
          Back to development

=== release 1.3.2 ===

2014-05-21 13:06:36 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * common:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.3.2

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

=== release 1.3.1 ===

2014-05-03 18:40:24 +0200  Sebastian Dröge <sebastian@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-rtsp-server.doap:
          Release 1.3.1

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>

        * 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:
          rtsp-server: sprinkle some allow-none annotations for g-i

2013-11-18 11:18:15 +0100  Wim Taymans <wim.taymans@gmail.com>

        * gst/rtsp-server/rtsp-stream.c:
        * gst/rtsp-server/rtsp-stream.h:
          stream: add method to filter transports
          Add a method to safely iterate and collect the stream transports
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664

2013-11-15 16:35:05 +0100  Wim Taymans <wim.taymans@gmail.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:
          rtsp: allow NULL func in filters
          Passing a null function make the filters return a list of
          refcounted objects.

2013-11-12 16:52:35 +0100  Wim Taymans <wim.taymans@gmail.com>

        * gst/rtsp-server/rtsp-address-pool.c:
        * tests/check/gst/addresspool.c:
          address-pool: fix address increment
          Use a guint instead of guint8 to increment the address. It's still not
          completely correct because a guint might not be able to hold the complete
          address range, but that's an enhacement for later.
          Add unit test to test improved behaviour.
          https://bugzilla.gnome.org/show_bug.cgi?id=708237

2013-11-12 10:55:14 +0100  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * tests/check/gst/client.c:
          client: allow absolute path in requests
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689

2013-11-07 13:22:09 +0100  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          client: make make_path_from_uri a vmethod

2013-11-12 12:04:55 +0100  Wim Taymans <wim.taymans@gmail.com>

        * docs/libs/gst-rtsp-server-sections.txt:
        * gst/rtsp-server/rtsp-stream.c:
        * gst/rtsp-server/rtsp-stream.h:
        * tests/check/Makefile.am:
        * tests/check/gst/stream.c:
          stream: Add functions to get rtp and rtcp sockets
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100

2013-11-12 11:21:55 +0100  Wim Taymans <wim.taymans@gmail.com>

        * gst/rtsp-server/rtsp-context.c:
        * gst/rtsp-server/rtsp-context.h:
          context: defing a GType for the context
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018

2013-10-12 23:56:00 +0200  Sebastian Pölsterl <sebp@k-d-w.org>

        * gst/rtsp-server/Makefile.am:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-context.c:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-mount-points.c:
        * gst/rtsp-server/rtsp-server.h:
        * gst/rtsp-server/rtsp-session-media.c:
        * gst/rtsp-server/rtsp-session.c:
        * gst/rtsp-server/rtsp-stream.c:
          Fixed several GIR warnings

2013-11-12 11:15:46 +0100  Wim Taymans <wim.taymans@gmail.com>

        * gst/rtsp-server/rtsp-auth.c:
          auth: small typos

2013-10-19 19:25:27 +0200  Sebastian Rasmussen <sebras@hotmail.com>

        * tests/check/Makefile.am:
        * tests/check/gst/token.c:
          tests: Add unit tests for token
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520

2013-10-19 19:24:34 +0200  Sebastian Rasmussen <sebras@hotmail.com>

        * 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

2013-10-19 19:21:53 +0200  Sebastian Rasmussen <sebras@hotmail.com>

        * 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

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>

        * docs/libs/gst-rtsp-server-sections.txt:
        * gst/rtsp-server/rtsp-media.c:
          media: add docs for new method

2013-07-02 18:55:28 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          Add API to rtsp-media set the pipeline's state

2013-06-11 19:09:42 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          Update current position/duration when gst_rtsp_media_get_range_string is called

2013-07-22 17:27:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-cgroups.c:
          tests: add some more docs

2013-07-22 14:25:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-cgroups.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-context.c:
        * gst/rtsp-server/rtsp-context.h:
        * gst/rtsp-server/rtsp-params.c:
        * gst/rtsp-server/rtsp-params.h:
        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-thread-pool.c:
        * gst/rtsp-server/rtsp-thread-pool.h:
        * tests/check/gst/client.c:
          ClientState -> Context
          Rename the clientstate to context and put the code in a separate file.

2013-07-18 12:19:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-auth.h:
          auth: add support for default token
          The default token is used when the user is not authenticated and can be used to
          give minimal permissions.

2013-07-18 11:44:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.c:
        * gst/rtsp-server/rtsp-auth.c:
          auth: use defines when possible

2013-07-18 11:44:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-address-pool.c:
          address-pool: improve docs

2013-07-18 12:26:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-permissions.c:
          permissions: add the role to the copy

2013-07-17 19:35:33 -0400  Olivier Crête <olivier.crete@collabora.com>

        * gst/rtsp-server/rtsp-permissions.c:
          permissions: Also copy the roles

2013-07-17 19:32:09 -0400  Olivier Crête <olivier.crete@collabora.com>

        * gst/rtsp-server/rtsp-permissions.c:
          permissions: Make it build

2013-07-16 12:36:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2013-07-12 14:10:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-rtsp-server-sections.txt:
        * gst/rtsp-server/rtsp-server.h:
          docs: fix docs

2013-07-12 12:41:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2013-07-12 12:38:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          client: add state push/pop

2013-07-12 12:36:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          client: add connection to state

2013-07-11 20:45:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-mount-points.c:
          mount-points: fix debug

2013-07-11 17:28:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/gst/media.c:
          tests: fix media test

2013-07-11 17:28:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-thread-pool.c:
          thread-pool: we don't require a state

2013-07-11 17:18:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-server.c:
          server: let context ref the server
          So that we don't risk losing the server object early anc crash.

2013-07-11 17:05:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/gst/client.c:
          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>

        * examples/test-auth.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-auth.h:
        * gst/rtsp-server/rtsp-client.c:
          auth: add auth checks
          Add an enum with auth checks and implement the checks in the auth object.
          Perform the checks from the client.

2013-07-05 20:48:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-auth.h:
        * gst/rtsp-server/rtsp-client.h:
          auth: use the token after authentication
          After we authenticated a user, keep the Token around in the state.

2013-07-05 20:43:39 +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:
        * tests/check/gst/media.c:
          media: add optional context for bus messages
          Add an optional mainloop to _prepare that will handle the bus messages instead
          of always using the shared mainloop.

2013-07-05 20:34:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/Makefile.am:
        * gst/rtsp-server/rtsp-token.c:
        * gst/rtsp-server/rtsp-token.h:
          token: add authorization token
          Add a simply miniobject that contains the authorizations. The object contains a
          GstStructure that hold all authorization fields. When a user is authenticated,
          the auth module will create a Token for the user. The token is then used to
          check what operations the user is allowed to do and various other configuration
          values.

2013-07-05 12:08:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.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.c:
        * gst/rtsp-server/rtsp-media-factory.h:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          auth: remove auth from media and factory
          Remove the auth object from media and factory. We want to have the RTSPClient
          authenticate and authorize resources, there is no need to place another auth
          manager on the media/factory.

2013-07-04 14:33:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-auth.h:
        * 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
          capabilities are allowed.

2013-07-03 16:15:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: error out on non-aggregate control
          We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.

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.

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()

2013-06-10 17:32:12 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          Add gst_rtsp_client_send_request API
          This makes it possible to send arbitrary messages to a client, such as
          SET_PARAMETER or GET_PARAMETER

2013-06-24 23:56:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: add _get_element() method
          Add method to get the element used when creating the media.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008

2013-06-24 23:51:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: fix docs

2013-06-24 11:41:27 -0700  Aleix Conchillo Flaque <aleix@oblong.com>

        * gst/rtsp-server/rtsp-stream.c:
        * gst/rtsp-server/rtsp-stream.h:
          stream: allow access to the rtp session
          https://bugzilla.gnome.org/show_bug.cgi?id=703004

2013-06-24 10:43:59 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-stream.c:
        * gst/rtsp-server/rtsp-stream.h:
          dscp qos support in gst-rtsp-stream
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645

2013-06-20 17:30:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/gst/rtspserver.c:
          tests: fix test
          Actually do what the comment says. Also keep the old code around, not sure what
          should happen when you get a 454 from a TEARDOWN, does it close the connection?
          it currently doesn't.

2013-06-20 12:20:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: also watch newly created session
          When we newly created a session, start watching it immediately instead of
          on the next request.

2013-06-20 12:18:23 +0200  Patricia Muscalu <patricia@axis.com>

        * tests/check/gst/client.c:
          tests: add unit test for new-session
          See https://bugzilla.gnome.org/show_bug.cgi?id=701587

2013-06-20 12:16:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: emit new-session when new session is created
          Only emit new-session when we created a new session for a client, not when a
          client picked up a previous session.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587

2013-06-20 11:17:29 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-client.c:
          client: handle asterisk as path in requests
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266

2013-06-20 11:14:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: handle segment query format mismatch
          It's possible that the segment query returns with a different format than what
          we asked for, handle this case also.

2013-06-11 15:28:32 +0200  David Svensson Fors <davidsf@axis.com>

        * gst/rtsp-server/rtsp-media.c:
          media: use segment stop in collect_media_stats
          Use segment stop instead of duration as range end point.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185

2013-06-17 16:47:56 +0200  Ognyan Tonchev <ognyan@axis.com>

        * gst/rtsp-server/rtsp-media.c:
        * tests/check/gst/media.c:
          rtsp-media: Do not leak the element in take_pipeline
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470

2013-06-17 16:18:37 +0200  Ognyan Tonchev <ognyan@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          rtsp-client: Make configure_client_transport virtual
          This patch makes configure_client_transport virtual. The functionality is
          needed to handle some weird clients sending multicast transport settings as url
          options.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173

2013-06-12 12:23:56 +0200  Ognyan Tonchev <ognyan@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          rtsp-client: Make param_set and param_get virtual
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072

2013-06-05 15:49:45 +0200  David Svensson Fors <davidsf@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: convert_range replaces get_range_times
          get_range_times worked for handling UTC ranges for seeks, but we also
          need to convert back from NPT to the requested unit in
          get_range_string. convert_range is now used for both.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084

2013-06-14 16:05:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-sdp.c:
        * gst/rtsp-server/rtsp-sdp.h:
          sdp: cleanup sdp info
          We don't need to pass the proto, we can more easily check a boolean.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063

2013-06-12 15:22:57 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-sdp.c:
          use 0.0.0.0 or :: for c= line instead of server address

2013-06-12 10:56:16 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-client.c:
          use local address, not remote, in SDP
          See https://bugzilla.gnome.org/show_bug.cgi?id=702063

2013-06-05 15:18:26 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 098c0d7 to 01a7a46

2013-05-29 13:45:00 +0200  David Svensson Fors <davidsf@axis.com>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: possibility to override range time conversion
          Make it possible to override the conversion from GstRTSPTimeRange to
          GstClockTimes, that is done before seeking on the media
          pipeline. Overriding can be useful for UTC ranges, where the default
          conversion gives nanoseconds since 1900.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191

2013-06-03 12:04:44 +0200  Ognyan Tonchev <ognyan@axis.com>

        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-server.h:
          rtsp-server: Expose the use_client_settings API
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935

2013-05-30 08:07:48 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-stream.c:
        * gst/rtsp-server/rtsp-stream.h:
          rtspstream: handle both ipv4 and ipv6 clients
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129

2013-05-31 15:28:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-sdp.c:
          Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
          This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
          We already have a way to place extra attributes in the SDP by using a string
          property with prefix x- or a- in the caps.

2013-05-31 15:27:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-sdp.c:
          Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
          This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
          We already have a way to place extra attributes in the SDP, just make a string
          property in the payloader with a- or x- prefix.

2013-05-31 15:41:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-sdp.c:
          rtsp: place a- and x- properties as attributes
          application/x-rtp has properties with a- and x- prefixes that should be
          placed as attributes in the SDP for the media instead of being added to the
          fmtp.

2013-05-31 12:10:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/Makefile.am:
        * examples/test-video.c:
          example: add TLS example

2013-05-31 11:42:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-server.h:
          server: add support for TLS
          Add methods to set and get a TLS certificate.
          Add vmethod to configure a new connection. By default, configure the TLS
          certificate in a new connection if needed.

2013-05-31 11:14:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-server.h:
          server: remove accept_client vmethod
          This vmethod is not very useful so remove it.

2013-05-30 17:23:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-server.c:
          server: don't crash on NULL GError

2013-05-30 10:46:33 +0200  Patricia Muscalu <patricia@axis.com>

        * gst/rtsp-server/rtsp-session-pool.c:
          rtsp-session-pool: corrected session timeout detection
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253

2013-05-30 10:52:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: improve debug

2013-05-30 07:18:22 +0200  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: refactor connection setup
          Let the server accept the socket connection and construct a GstRTSPConnection
          from it. Remove the code from the client and let the client only deal with
          a fully configure GstRTSPConnection object.
          We will need this later when the server will configure the connection for
          TLS.

2013-05-30 06:49:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-stream.c:
          stream: keep the transport object alive
          Keep the transport object alive while we have it as qdata on the
          source.

2013-05-27 12:58:07 +0200  Alexander Schrab <alexas@axis.com>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-server.c:
          rtsp-server: Do not crash on nmapping of server
          * 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

2013-02-19 16:43:08 -0500  Olivier Crête <olivier.crete@collabora.com>

        * gst/rtsp-server/rtsp-stream.c:
          stream: Properties are always there in Gst 1.0

2013-02-19 16:36:20 -0500  Olivier Crête <olivier.crete@collabora.com>

        * tests/check/gst/addresspool.c:
          tests: Add tests for unicast addresses in pool

2013-02-20 14:26:03 -0500  Olivier Crête <olivier.crete@collabora.com>

        * 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

2013-02-19 16:34:16 -0500  Olivier Crête <olivier.crete@collabora.com>

        * 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:
          address-pool: Add unicast addresses

2013-02-19 13:19:41 -0500  Olivier Crête <olivier.crete@collabora.com>

        * configure.ac:
        * 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
          threads.

2013-02-19 12:31:23 -0500  Olivier Crête <olivier.crete@collabora.com>

        * gst/rtsp-server/rtsp-server.c:
          rtsp-server: No need to store the GMainContext in the client context

2013-02-18 20:22:18 -0500  Olivier Crête <olivier.crete@collabora.com>

        * tests/check/gst/rtspserver.c:
          tests: Add test for client disconnection

2013-02-18 20:15:41 -0500  Olivier Crête <olivier.crete@collabora.com>

        * tests/check/gst/rtspserver.c:
          tests: Test client and session timeouts with multiple threads

2013-02-18 14:59:58 -0500  Olivier Crête <olivier.crete@collabora.com>

        * gst/rtsp-server/rtsp-address-pool.c:
        * gst/rtsp-server/rtsp-auth.c:
        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-media-factory-uri.c:
        * gst/rtsp-server/rtsp-media-factory.c:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-mount-points.c:
        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-session-media.c:
        * gst/rtsp-server/rtsp-session-pool.c:
        * gst/rtsp-server/rtsp-session.c:
          Document locking and its order

2013-02-15 20:02:31 -0500  Olivier Crête <olivier.crete@collabora.com>

        * tests/check/gst/rtspserver.c:
          tests: Test that slow DESCRIBE don't block other clients

2013-02-14 19:52:09 -0500  Olivier Crête <olivier.crete@collabora.com>

        * tests/check/gst/client.c:
          tests: Add tests for client-requested multicast address

2013-02-14 13:44:54 -0500  Olivier Crête <olivier.crete@collabora.com>

        * docs/libs/gst-rtsp-server-sections.txt:
          docs: Put the various functions in the right sections

2013-02-14 13:38:07 -0500  Olivier Crête <olivier.crete@collabora.com>

        * 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

2012-11-29 17:20:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: require an element in media constructor

2012-11-29 17:07:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          Revert "client: TEARDOWN brings that state to Init again"
          This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
          The object is already disposed, there is no point in setting the state.

2012-11-29 12:30:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: TEARDOWN brings that state to Init again

2012-11-29 11:11:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-rtsp-server-sections.txt:
        * examples/test-auth.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.c:
        * gst/rtsp-server/rtsp-media.h:
        * gst/rtsp-server/rtsp-mount-points.c:
        * gst/rtsp-server/rtsp-mount-points.h:
        * 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-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/media.c:
          rtsp: make object details private
          Make all object details private
          Add methods to access private bits

2012-11-28 14:50:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/Makefile.am:
        * tests/check/gst/media.c:
          tests: add media tests

2012-11-28 14:45:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: check if prepared for some methods
          Check that the media object is prepared before doing seek and getting the
          current position etc.
          Add some g_return checks.

2012-11-28 12:40:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/Makefile.am:
        * tests/check/gst/mediafactory.c:
          tests: add mediafactory test

2012-11-28 12:40:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-stream.c:
          stream: improve debug

2012-11-28 12:39:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: unref pipeline in finalize to avoid leaking it

2012-11-28 12:10:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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:
        * 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
          Describes better what the object manages.

2012-11-26 09:36:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * configure.ac:
          configure: bump required version of -base

2012-11-21 17:21:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: fix seeking

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.

2012-11-21 16:41:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-launch.c:
          examples: fix whitespace

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

2012-10-27 21:05:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          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
          stream over UDP.

2012-10-27 11:53:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-mp4.c:
        * gst/rtsp-server/rtsp-media.c:
          small fixes to docs and debug

2012-10-26 17:29:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-stream.c:
          stream: transports must already have been removed

2012-10-26 17:28:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * 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
          Add some docs

2012-10-26 15:23:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: move unprepare below default implementation
          Makes it easier to find the default implementation

2012-10-26 15:21:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: signal unprepared when we actually finish

2012-10-26 15:19:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: no need to unlock, unprepare does that when needed

2012-10-26 12:33:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2012-10-26 12:04:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * 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>

        * common:
          Automatic update of common submodule
          From 7fda524 to 464fe15

2012-04-04 14:45:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * configure.ac:
        * docs/libs/Makefile.am:
        * docs/version.entities.in:
        * gst-rtsp.spec.in:
        * gst/rtsp-server/Makefile.am:
        * pkgconfig/Makefile.am:
        * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtsp-server.pc.in:
        * tests/Makefile.am:
          rtsp-server: Update versioning

2012-03-29 15:12:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

          Merge remote-tracking branch 'origin/0.10'
          Conflicts:
          gst/rtsp-server/rtsp-session-pool.c

2012-03-27 10:13:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/rtsp-server/rtsp-session-pool.c:
          rtsp-server: Don't use deprecated GLib API

2012-03-26 12:23:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

          Replace master with 0.11

2012-03-26 12:22:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11

2012-03-26 12:20:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11

2012-03-19 10:48:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

        * docs/README:
          A couple minor typo fixes

2012-03-13 18:10:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
          media: fix state of the appqueue

2012-03-13 16:06:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media-factory-uri.c:
          factory: use videoconvert

2012-03-13 16:02:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media-factory-uri.c:
          factory: change to new style caps

2012-03-07 15:03:55 +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-factory-uri.c:
        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-server.c:
        * gst/rtsp-server/rtsp-server.h:
        * gst/rtsp-server/rtsp-session-pool.c:
          rtsp-server: port to GIO
          Port to GIO

2012-03-07 15:03:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * configure.ac:
          configure: fix build

2012-02-29 15:56:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * docs/README:
          docs: fix for gst_rtsp_server_set_port() -> _set_service()
          https://bugzilla.gnome.org/show_bug.cgi?id=666548

2012-02-13 11:42:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
        * examples/Makefile.am:
          First rule of gst-rtsp-server club: don't talk about gst-phonon

2012-02-13 11:40:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
        * pkgconfig/Makefile.am:
        * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtsp-server.pc.in:
          pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
          For consistency with all other modules.

2012-02-13 11:06:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          rtsp-client: update for new map API

2012-02-13 10:37:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * .gitignore:
        * bindings/Makefile.am:
        * 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:
        * bindings/python/test.py:
        * configure.ac:
          python: remove pygst-based python bindings
          pygi is the future, apparently.

2012-01-25 14:12:41 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>

        * 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

2012-01-18 16:48:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 0807187 to 2a59016

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

2011-12-09 11:00:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-auth.c:
          example: update for new caps

2011-12-09 10:53:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-video.c:
        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-media-factory-uri.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: port some more to 0.11
          Fix caps.
          Remove bufferlist stuff
          Update for new API.
          Add queue before appsink now that preroll-queue-len is gone.
          Update for request pad changes.

2011-11-03 16:14:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11

2011-11-03 16:06:23 +0100  Fabian Deutsch <fabian.deutsch@gmx.de>

        * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
          bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
          Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>

2011-11-03 16:06:23 +0100  Fabian Deutsch <fabian.deutsch@gmx.de>

        * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
          bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
          Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>

2011-11-03 12:58:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11

2011-11-03 12:55:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: add a seekable boolean
          Maintain the seekable state with a new variable instead of reusing the
          is_live variable.

2011-09-16 11:31:17 -0400  Victor Gottardi <vgottardi@hotmail.com>

        * gst/rtsp-server/rtsp-media.c:
          Disallow seek in live media

2011-11-03 11:58:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11

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>

        * common:
          add common

2011-04-26 19:07:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' into 0.11
          Conflicts:
          common
          configure.ac

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

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

2011-04-20 11:13:56 +0200  Alessandro Decina <alessandro.d@gmail.com>

        * bindings/python/arg-types.py:
          python bindings: add GstRTSPUrlParam
          Needed to implement MediaFactory virtual proxies

2011-04-20 10:19:46 +0200  Alessandro Decina <alessandro.d@gmail.com>

        * bindings/python/arg-types.py:
          python bindings: fix returning GstRTSPUrl types

2011-04-20 10:17:07 +0200  Alessandro Decina <alessandro.d@gmail.com>

        * bindings/python/arg-types.py:
          python bindings: add arg type for GstRTSPUrl

2011-04-20 10:16:08 +0200  Alessandro Decina <alessandro.d@gmail.com>

        * bindings/python/rtspserver.defs:
          python bindings: fix the definition of MediaFactory.collect_stream

2011-04-04 15:59:50 +0300  Stefan Kost <ensonic@users.sf.net>

        * 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>

        * 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
          fire the 'closed' signal.

2011-01-12 15:35:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-client.h:
          client: emit signal when closing

2011-01-12 13:57:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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
          permissions when accessing the factory.
          Add hints to the auth methods for future more fine grained authorisation.
          Add example application for per factory authentication.

2011-01-12 13:16:08 +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:
        * 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.

2011-01-12 12:07:40 +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 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.

2010-03-16 18:34:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-session.c:
          session: indent

2010-03-16 18:33:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: use right size for malloc

2010-03-10 11:45:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-server.c:
          server: comment ipv6 server listening address

2010-03-10 11:45:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

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:
          Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.

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

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

2009-06-12 22:22:40 +0200  Sebastian <sebastian@ubuntu.(none)>

        * gst/rtsp-server/rtsp-media.c:
          Don't use name for gstrtpbin so we can add multiple instances to the pipeline

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

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

2009-04-17 21:13:07 +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

2009-04-14 23:38:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/Makefile.am:
        * gst/rtsp-server/rtsp-client.c:
        * gst/rtsp-server/rtsp-media.c:
          server: use appsink and appsrc with the API
          Use the appsink/appsrc API instead of the signals for higher
          performance.

2009-04-14 23:38:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-ogg.c:
          tests: set the payload type correctly

2009-04-03 22:46:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media-factory.c:
          factory: connect to the unprepare signal
          Connect to the unprepare signal for non-reusable media so that we can remove
          them from the cache.

2009-04-03 22:45:57 +0200  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 unprepare

2009-04-03 22:22:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          media: more work on making the media shared
          Add a reusable flag to medias, indicating that they can be reused after a state
          change to NULL.
          Small cleanups.

2009-04-03 19:47:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * examples/test-readme.c:
          examples: mark the example as shared for testing

2009-04-03 19:44:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-media.c:
        * gst/rtsp-server/rtsp-media.h:
          client: support shared media
          Always perform the state actions even if the target state of the pipeline is
          already correct, we still want to add/remove the transports when we are dealing
          with shared media.
          Keep a counter of the number of active transports for a media so that we can use
          this to perform a state change when needed.
          Perform a state change of the pipeline only when the first transport was added
          or when there are no active transports.

2009-04-03 09:03:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          client: fix refcounting crasher
          Don't need to remove the weak refs in the finalize methods, they are already
          removed in the dispose.
          Don't register the callback with a DestroyNofity.

2009-04-01 01:01:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/rtsp-server/rtsp-client.c:
          Fix rtsp client refcount management in TCP mode.
          Don't unref a client ref we never had. Fixes an unref
          of an already-free client object after a client
          teardown request for me.

2009-04-01 00:45:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/rtsp-server/rtsp-session.c:
          docs: fix typo in API docs

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.
          Add some more debug info.
          Only seek when we need to.
          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
          Add a timeout property ot the client and make the other properties into GObject
          properties.

2009-02-10 16:21:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/rtsp-server/rtsp-session-pool.c:
          Use getters and setters in property code
          Use the getters and setters for the timeout property instead of locking
          ourselves.

2009-02-04 20:13:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server

2009-02-04 20:10:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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:
          Add more timeout stuff
          Add method to check if a session is expired.
          Add method to perform cleanup on a session pool.

2009-02-04 19:52:50 +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-pool.h:
        * gst/rtsp-server/rtsp-session.c:
        * gst/rtsp-server/rtsp-session.h:
          Add beginnings of session timeouts and limits
          Add the timeout value to the Session header for unusual timeout values.
          Allow us to configure a limit to the amount of active sessions in a pool. Set a
          limit on the amount of retry we do after a sessionid collision.
          Add properties to the sessionid and the timeout of a session. Keep track of
          creation time and last access time for sessions.

2009-02-04 17:00:42 +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:
        * gst/rtsp-server/rtsp-session-pool.c:
        * gst/rtsp-server/rtsp-session.c:
        * gst/rtsp-server/rtsp-session.h:
          Cleanup of sessions and more
          Fix the refcounting of media and sessions in the client. Properly clean up the
          session data when the client performs a teardown.
          Add Server header to responses.
          Allow for multiple uri setups in one session.
          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:
        * examples/Makefile.am:
        * examples/main.c:
        * gst/Makefile.am:
        * gst/rtsp-server/Makefile.am:
        * 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-server.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:
        * src/Makefile.am:
          Split in library and example program

2008-11-10 20:59:35 +0100  Sebastian Pölsterl <sebp@ubuntu.(none)>

        * src/rtsp-client.h:
          Removed obsolete variable

2008-11-10 21:03:15 +0100  Sebastian Pölsterl <sebp@ubuntu.(none)>

        * src/rtsp-client.c:
        * src/rtsp-client.h:
          Removed pipeline variable GstRTSPClient, because it's only used in one function

2009-01-08 11:22:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * src/rtsp-media.c:
          Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.

2008-10-23 12:23:27 +0200  Wim Taymans <wim@metal.(none)>

        * src/rtsp-session.c:
          Initialize some more vars.

2008-10-23 12:14:55 +0200  Wim Taymans <wim@metal.(none)>

        * src/rtsp-session.c:
          Initialize variable to avoid compiler warning.

2008-10-09 13:30:47 +0100  Simon McVittie <simon.mcvittie@collabora.co.uk>

        * .gitignore:
          Add a reasonable generic .gitignore