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38 commits

Author SHA1 Message Date
Wim Taymans
8d439edd7a rtspconnection: add flush method 
Add a method to set/unset the flushing state that makes _wait_backlog()
unlock.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
 2014-03-28 09:34:33 +01:00
Göran Jönsson
0b30fdbfbe 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
 2014-03-10 17:28:40 +01:00
David Svensson Fors
6cd0d10d30 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
 2014-03-10 10:43:03 +01:00
Aleix Conchillo Flaqué
0a115bd31f rtspconnection: allow specifying a certificate database 
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
 2014-02-19 21:48:13 +01:00
Sebastian Dröge
b0aad9dd84 rtspconnection: Fix indention in header 2013-11-01 16:43:56 +01:00
Aleix Conchillo Flaque
53c7ad0c87 rtspconnection: allow setting tls certificate validation 
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
 2013-11-01 16:42:34 +01:00
Wim Taymans
0b933ff87b rtsp: add method to get the TLS connection 2013-05-30 17:31:13 +02:00
Olivier Crête
aef8de337c rtspconnection: Add API to disable session ID caching in the connection 
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()
 2013-03-11 10:41:00 +01:00
Wim Taymans
65c5ecd270 rtspconnection: add limit to queued messages 
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()
 2012-12-14 11:36:58 +01:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Sebastian Dröge
aed2666b53 rtsp: Port to GIO 2012-01-17 16:38:45 +01:00
Wim Taymans
bdf3845498 rtsp: cleanup headers 
Add padding, fix indentation, remove deprecated stuff
 2011-11-11 19:35:33 +01:00
Wim Taymans
318fbf3310 rtspconnection: Handle closed POST socket in tunneling 
Catch more socket errors.
Rework how sockets are managed in the GSource, wake up the maincontext instead
of adding/removing the sockets from the source.
Add callback for when the tunnel connection is lost. Some clients (Quicktime
Player) close the POST connection in tunneled mode and reopen the socket when
needed.

See #612915
 2010-04-06 10:59:39 +02:00
Peter Kjellerstedt
2882c22d95 rtsp: Added missing Since tags. 2009-08-24 13:58:50 +02:00
Peter Kjellerstedt
066f9be5c9 rtsp: Added new API for sending using GstRTSPWatch. 
The new API to send messages using GstRTSPWatch will first try to send the
message immediately. Then, if that failed (or the message was not sent
fully), it will queue the remaining message for later delivery. This avoids
unnecessary context switches, and makes it possible to keep track of
whether the connection is blocked (the unblocking of the connection is
indicated by the reception of the message_sent signal).

This also deprecates the old API (gst_rtsp_watch_queue_data() and
gst_rtsp_watch_queue_message().)

API: gst_rtsp_watch_write_data()
API: gst_rtsp_watch_send_message()
 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
fb3b761af5 rtsp: Added gst_rtsp_connection_set_http_mode(). 
With gst_rtsp_connection_set_http_mode() it is possible to tell the
connection whether to allow HTTP messages to be supported. By enabling HTTP
support the automatic HTTP tunnel support will also be disabled.

API: gst_rtsp_connection_set_http_mode()
 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
e5ec74c7a9 rtsp: Add error_full callback to GstRTSPWatchFuncs. 
The error_full callback is similar to the error callback, but allows for
better error handling. For read errors a partial message is provided to
help an RTSP server generate a more correct error response, and for write
errors the write queue id of the failed message is returned.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
dd7d0cfc45 rtsp: Added gst_rtsp_connection_create_from_fd(). 
API: gst_rtsp_connection_create_from_fd()
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
726a47f777 rtsp: Added gst_rtsp_watch_queue_data(). 
gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
but allows for queuing any data block for writing (much like
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)

API: gst_rtsp_watch_queue_data()
 2009-06-17 18:34:33 +02:00
Peter Kjellerstedt
ff38999c8b rtsp: Call message_sent() callback for all sent messages. 
Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
 2009-06-15 09:28:13 +02:00
Wim Taymans
59d9833924 rtsp: add G_LIKELY because we can 2009-06-02 12:10:39 +02:00
Peter Kjellerstedt
848b834cb9 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined. 2009-06-01 09:58:27 +02:00
Wim Taymans
8b37dc3eb8 rtsp: add support for proxies 
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
 2009-03-31 19:00:00 +02:00
Wim Taymans
2cc1a6808d rtsp: Add methods for getting the read/write fds 
API:gst_rtsp_connection_get_readfd()
API:gst_rtsp_connection_get_writefd()
 2009-03-06 18:54:57 +01:00
Wim Taymans
629f2dcee4 rtsp: fix the memory management of the url 
Constify the url parameter in _create.
Make a copy of the url stored in the connection.
Free the url when the connection is freed.
 2009-03-04 16:11:20 +01:00
Wim Taymans
b6d7a1dc03 RTSP: Add support for server tunneling 
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()
 2009-03-04 12:21:29 +01:00
Wim Taymans
fbc4f2d4fe RTSP: add support for Quicktime tunneled RTSP 
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()
 2009-03-02 16:03:49 +01:00
Wim Taymans
4664fe40bc rtsp: add _get_url method and separate sockets 
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()
 2009-03-02 10:58:49 +01:00
Wim Taymans
f187ffddce Make RTSPConnection opaque and rename RTSPChannel 
Make the RTSPConnection object opaque so that we can extend it in the future.

Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
 2009-02-19 15:55:07 +01:00
Wim Taymans
a2f04c8f61 Add RTSP accept method 
Add a method to accept a connection on a socket and create a GstRTSPConnection
for it.

API: gst_rtsp_connection_accept()
 2009-02-18 18:46:35 +01:00
Wim Taymans
a6d75bd33c Add RTSP channel object for async io 
Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
that the connection can be monitored from a maincontext. This allows us to
operate in ASYNC mode, which is handy when building a server.

Rework the old code to use the async code under the hood.

API: gst_rtsp_channel_new()
API: gst_rtsp_channel_unref()
API: gst_rtsp_channel_attach()
API: gst_rtsp_channel_queue_message()
 2009-02-18 17:42:59 +01:00
Peter Kjellerstedt
26cd5ea1c8 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards. 
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.
 2008-06-04 11:53:53 +00:00
Wouter Cloetens
a8a2b9c717 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065. 
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
 2008-05-08 14:46:27 +00:00
Wim Taymans
0667334b2d gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection. 
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
(gst_rtsp_connection_flush):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Use GstPoll for the rtsp connection.
 2008-02-28 09:50:52 +00:00
Wim Taymans
01d9553d43 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups. 
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Small cleanups.
On shutdown, don't read the control socket yet.
Set timeout value correctly in all cases.
Add function to check if the server accepts reads or writes.
API: gst_rtsp_connection_poll()
* gst-libs/gst/rtsp/gstrtspdefs.h:
Fix compilation with -pedantic.
Add enum for _poll.
 2007-08-17 13:42:49 +00:00
Wim Taymans
3dff14d6b1 Added beginnings of RTSP documentation. 
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (add_auth_header),
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
(read_body), (gst_rtsp_connection_receive),
(gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtspurl.h:
Added beginnings of RTSP documentation.
 2007-07-24 19:19:33 +00:00
Wim Taymans
19e0dd3140 Move SDP and RTSP from helper objects in -good to a reusable library. 
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
(parse_response_status), (parse_request_line), (parse_line),
(gst_rtsp_connection_read), (read_body),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
(gst_rtsp_find_method):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_new), (gst_rtsp_message_init),
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
(gst_rtsp_message_dump):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse), (gst_rtsp_range_free):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
(range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
(gst_rtsp_transport_free):
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
(gst_sdp_attribute_init), (gst_sdp_message_new),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
(gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
(read_string), (read_string_del), (gst_sdp_parse_line),
(gst_sdp_message_parse_buffer), (print_media),
(gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Move SDP and RTSP from helper objects in -good to a reusable library.
Use a proper gst_ namespace.
 2007-07-24 11:52:56 +00:00