Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
which fixes the build of pbutils on windows (#574319).

Disconnect the notify::caps signal in our callback (it'll be re-added
if we're not, in fact, finished getting complete caps). Ensures that
caps changes mid-stream (e.g. from an mp3 that changes from
stereo->mono mid-file) don't cause us to try to add a new pad.

--

Add some more sprinle examples and add some more comments.
See #574160.

--

For unsigned int out+in can't be < 0.

Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
unconnected state as it was allocated.

Constify the url parameter in _create.
Make a copy of the url stored in the connection.
Free the url when the connection is freed.

Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
that a server can store and match the id against other tunnel requests.

Fix the URI in the tunnel requests so that they contain the absolute uri and the
query string if any instead of just the hostname.

Transparently base64 decode the input stream when tunneling.

Add method to set the connection ip address so that it can be included in the
tunnel response.

Add method to connect the two tunnel requests.

Add two callbacks for the async mode to notify a tunnel start and tunnel
complete event.

Add method to reset the watch after the connection has been tunneled.

Various little refactoring to make more stuff reusable.

API: RTSP::gst_rtsp_connection_set_ip()
API: RTSP::gst_rtsp_connection_get_tunnelid()
API: RTSP::gst_rtsp_connection_do_tunnel()
API: RTSP::gst_rtsp_watch_reset()

Add two more result codes for tunneling support.

Remove , from last enum member to improve compatibility with other compilers.

Fixes: #572993.  Patch author prefers to use an alias, contact
ds if you actually need a real name.

Signed-off-by: David Schleef <ds@schleef.org>

--

Add support for tunneling RTSP over HTTP.
Fix documentation some more.
See also #573173.

API: RTSP:gst_rtsp_connection_is_tunneled()
API: RTSP:gst_rtsp_connection_set_tunneled()

Add transport define for RTSP tunneled over HTTP.

Parse rtsph:// uris as tunneled HTTP over TCP.

API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP

See also #573173.

No, I'm not wim's buildslave, seriously.

Add gst_rtsp_connection_get_url() method.

Reserve space for 2 sockets, one for reading and one for writing. Use socket
pointers to select the read and write sockets. This should allow us to implement
tunneling over HTTP soon.

API: RTSP::gst_rtsp_connection_get_url()

The previous change to appsrc/appsink requires people to 'make clean'
to get the marshallers rebuilt (causing a build failure otherwise).
Change some lines in the .list file around to force a rebuild of
these files automatically.

Fixes bug #573528.

Fixes bug #573529.

This comes mostly from a review of ffmpeg/libavformat/riff.c

Free the gstRTSPTimeRange if we don't return it. Also simplify
gst_rtsp_range_free() as it is valid to pass NULL to g_free().

We can break, once we have a name to make sure, we won't read it ever twice.

Add a uri handler to appsink.
don't emit signals when we have installed callbacks on appsink.

Add callbacks to appsrc to replace the signals.
Add property to disable callbacks in appsrc, default to TRUE for backwards
compatibility but disable when callbacks are installed.

API: GstAppSrc::emit-signals
API: GstAppSrc::gst_app_src_set_emit_signals()
API: GstAppSrc::gst_app_src_get_emit_signals()
API: GstAppSrc::gst_app_src_set_callbacks()

Add some padding to the callbacks structure just to be safe.

Remove the now invisible marshaller methods from the docs.

Fix a comment in the unit test.

Add appsrc/sink symbols to private, as they are covered in the libs docs.