rtspdefs: add RFC 4567 headers and status code

This new Header and status code is used for SRTP
rtspconnection: GstRTSPWatch func for tunnel GET response

Add a callback in GstRTSPWatch where the response to HTTP GET for
tunneled connections can be modified.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
rtspconnection: gst_rtsp_watch_wait_backlog

New method that wait until there is room in backlog queue.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
oggmux: create vp8 header data if not provided in caps

vp8 stream header shouldn't be assumed to be provided in caps always
as this would repeat the same code in all demuxers/encoders. Instead,
make oggmux generate them if they are not supplied.

https://bugzilla.gnome.org/show_bug.cgi?id=722682
oggmux: implement vp8 granulepos function

Add an extra function to the oggstream map to inform it about
the incoming buffers. This way oggmux can keep a count on the
vp8 invisible frames and calculate the granulepos correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=722682
Add new header file
adder: set a group-id on the stream-start event

Set a default group-id to fix a warning printed by the sink.
docs: update plugin docs and remove old properties and signals

Re-generate .args and .signals file from scratch so that
old signals that no longer exist (such as the 'new-decoded-pad'
signal on decodebin) no longer show up in the documentation.
playsink: Fix documentation for what the audio chain looks like

https://bugzilla.gnome.org/show_bug.cgi?id=679031
rtspconnection: Silence a compiler warning

Cast the argument into (const char *) on W32, as winsock2 expects it.

https://bugzilla.gnome.org/show_bug.cgi?id=726433
playback: Add video-/audio-filter properties

This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will be
applied if possible -- for non-raw sinks, the filters will be skipped.

If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.

https://bugzilla.gnome.org/show_bug.cgi?id=679031
Revert "playback: Add video-/audio-filter properties"

This reverts commit fb8fdedb4f4649aa33700bbc720131c1678df49f.
playback: Add video-/audio-filter properties

This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will
e
applied if possible -- for non-raw sinks, the filters will be skipped.

If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.

https://bugzilla.gnome.org/show_bug.cgi?id=679031
tests: decodebin: port old decodebin2 test for parser and decoder linking

They were in the old decodebin2.c tests file and were never ported.
Now we can get rid of decodebin2.c
rtpbasepayload: add pt and ssrc to stats
rtspconnection: Fix connection_poll()

* Only check for conditions we are interested in.
* Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
  will always be reported if they are true.
* Do not create timed source if timeout is NULL.
* Correctly wait for sources to be dispatched, context_iteration() is
  not guaranteed to always block even if set to do so.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
rtspconnection: Fix minor memory leaks in error handling

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
mikey: add MIKEY parsing helpers

MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
docs: add MIKEY docs
rtsptransport: UDP is also default for SAVP and AVPF