Properly handle protocol version in the connection
Add the following headers types:
* Pipelined-Request
* Media-Properties
* Seek-Style
* Accept-Ranges
https://bugzilla.gnome.org/show_bug.cgi?id=781446
This way special characters such as '@' can be used in
usernames or passwords, e.g.
rtsp://view:%40dm%4An@<IP-ADDR>/media/camera1
will now parse username and password into:
User: view
Pass: @dm:n
https://bugzilla.gnome.org/show_bug.cgi?id=758389
To make the structs usable in bindings, and fix
gstrtspmessage.c:1188: Warning: GstRtsp:
gst_rtsp_message_parse_auth_credentials: return value: Invalid
non-constant return of bare structure or union; register as
boxed type or (skip)
https://bugzilla.gnome.org/show_bug.cgi?id=774416
It is actually needed as we need some symbols. We do not link
to libgstsdp as the user of the lib should do it (same with
autotools build).
This reverts previous commit
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.
https://bugzilla.gnome.org/show_bug.cgi?id=758922
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled. The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.
The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.
https://bugzilla.gnome.org/show_bug.cgi?id=749596
Some servers incorrectly parse header names with strict case-sensitivity. For
compatibility with these systems change X-Sessioncookie to x-sessioncookie.
https://bugzilla.gnome.org/show_bug.cgi?id=758921
GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not
including the generated file makes older gst-p-good fail to build,
so it constitues an API break.
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."
https://bugzilla.gnome.org/show_bug.cgi?id=739132
Add API to add and get custom headers that are not
covered by our header fields enum. This is backwards
compatible in that it will also work for our defined
fields, so if we ever add a new header field to the
enum, get_header_by_name() for the same header string
will still work.
API: gst_rtsp_message_add_header_by_name()
API: gst_rtsp_message_take_header_by_name()
API: gst_rtsp_message_remove_header_by_name()
API: gst_rtsp_message_get_header_by_name()
The timeout parameter is only allowed in a session response header
but some clients, like Honeywell VMS applications, send it as part
of the session request header. Ignore everything from the semicolon
to the end of the line when parsing session id.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736267
Fixes a crash when controlsrc, readsrc or writesrc are modified from
gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
same time.
https://bugzilla.gnome.org/show_bug.cgi?id=735569
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
Add a method to make a media-type from the transport. Deprecate the old
method that only used the mode.
Based on patch from Aleix Conchillo Flaqué <aleix@oblong.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720219
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.
