Add a read source on write socket when lost tunnel.
To be able to detect when clint closes get channel.
This is already done in gst_rtsp_source_dispatch_write but
only when the queue is empty.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730368
By re-using the uri argument for storing local data, we could end up in
a situation where we would free uri ... which would actually be the
string passed in argument.
Instead explicitely use a local variable. Fixes double-free issues.
CID #1212176
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
* 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
Two new functions have been added,
gst_rtsp_connection_set_tls_database() and
gst_rtsp_connection_get_tls_database(). The certificate database will be
used when a certificate can't be verified with the default database.
https://bugzilla.gnome.org/show_bug.cgi?id=724393
Added new functions gst_rtsp_connection_set_tls_validation_flags() to
allow setting the TLS certificate validation flags when establishing a
TLS connection.
A getter is also available, gst_rtsp_connection_get_tls_validation_flags().
https://bugzilla.gnome.org/show_bug.cgi?id=711231
Creating a GSource and not attaching it to a context will cause
a leak of it's child sources. That is why we create writesrc right
before attaching it to a context.
https://bugzilla.gnome.org/show_bug.cgi?id=708667
When we start to read a message, we need to continue reading until the end of
the message or else we lose track and cause parse errors. Use a variable
may_cancel to avoid cancelation after we read the first byte until we have
the complete message.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088
Don't try to read/write from an inactive stream. When we, for example,
transfer the second connection in tunneling mode, we are not interested anymore
on read/write activity on the old connection.
Calculate the local IP address in the accept call. We need to place this IP
address in the GET reply in the X-Server-IP-Address header so that the client
knows where to send the POST to in case of tunneled RTSP. Before this patch
it used the client IP address, which would make the client send the POST request
to itself and fail.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697092
This is necessary to allow having more than one session in the same connection.
API: gst_rtsp_connection_set_remember_session_id()
API: gst_rtsp_connection_get_remember_session_id()
Add a limit to the amount of queued bytes or messages we allow on the watch.
API: GstRTSPConnection::gst_rtsp_watch_set_send_backlog()
API: GstRTSPConnection::gst_rtsp_watch_get_send_backlog()
