rtspsrc: add "proxy-id" and "proxy-pw" properties

to match souphttpsrc. user/password passed via the URI
will still take precedence though.

https://bugzilla.gnome.org/show_bug.cgi?id=395427
examples: check for uri argument in decodebin-h264p-amr server example

Otherwise people get a rather confusing error message.
cairo: port cairooverlay to 0.11

The other elements are not that interesting now that we're
using pangocairo in the pango plugin, and should probably
just be removed.
cairo: remove old cairo-based text renderering element

They haven't worked well or at all in a very long time
and were rather bit-rotten, and there's no need for them
any more.
docs: fix docs build and update after removal of old cairo elements
docs: update for new rtspsrc proxy-id and proxy-pw properties
souphttpsrc: error out properly when receiving data along with an error status

When receiving an error code from the http server, such as 404,
data might be sent along with it, like a web page. We don't want
to output that data in this case, and we also want to pass the
FLOW_ERROR return back to the base class, so it can stop properly.

https://bugzilla.gnome.org/show_bug.cgi?id=678429
tests: add test for souphttpsrc error handling with data

https://bugzilla.gnome.org/show_bug.cgi?id=678429
v4l2: Also handle the new ENOENT return value of VIDIOC_QUERYCTRL

https://bugzilla.gnome.org/show_bug.cgi?id=691098
udpsrc: sanity check size of available packet data for reading to avoid memory waste

On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.

https://bugzilla.gnome.org/show_bug.cgi?id=610364
rtpsession: improve debug
rtp: more debug
rtpsession: don't cast event functions

There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
rtp: include downstream latency in SR calculations

When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
jitterbuffer: refactor handle sync code

Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
rtp: small improvements
rtpsbcpay: Make a2dpsink to act like a bin and split the payloader.
rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps.
rtpsbcpay: Fix gtreamer payloader sending fragmented frames.
rtpsbcpay: Fixes gstreamer caps and code cleanup.