Commit graph

271 commits

Author SHA1 Message Date
Wim Taymans
d6579912cb rtspsrc: make setup url in a smarter way
Make sure we always separate the base and control url parts with a / when
creating the setup url.
2010-05-04 16:36:15 +02:00
Alessandro Decina
c8a02a91a6 rtspsrc: handle SEEKING queries. 2010-05-04 16:05:13 +02:00
Wim Taymans
b84bf10455 rtspsrc: add property to control the buffering method
Add a property to control how the jitterbuffer performs timestamping and
buffering.
2010-04-05 15:26:03 +02:00
Benjamin Otte
3f511ec361 Add -Wwrite-strings to the configure flags
... and fix all warnings
2010-03-21 14:17:47 +01:00
Wim Taymans
ef804589ca rtsp: use GType from -base and bump required version
Use the transport flags GType from -base and bump the required version of -base
because of this.
2010-03-19 15:03:43 +01:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Benjamin Otte
1055aaa9cb Add -Wredundant-decls warning flag
Also fix compile issues
2010-03-17 19:35:10 +01:00
Benjamin Otte
3342b1679e Add -Wmissing-declarations -Wmissing-prototypes warning flags
And fix all the warnings.
2010-03-17 18:23:28 +01:00
Wim Taymans
ba6dbaecfc rtspsrc: don't forget to send keepalive messages
When we operate in TCP mode, still send keepalive messages when we
need to.

Fixes #612696
2010-03-15 11:38:23 +01:00
Wim Taymans
d29fa60f97 rtspsrc: check for NULL before doing strcmp
Check the connection and address type for NULL before doing strcmp and
crashing.

Fixes #612553
2010-03-11 12:56:11 +01:00
Wim Taymans
821096c4f1 rtspsrc: parse connection information
Parse the connection information from the SDP and use it to figure out if we are
dealing with ipv4 or ipv6 connections.
2010-03-10 11:28:22 +01:00
Wim Taymans
8eb5c2c794 rtspsrc: require a destination for multicast
When setting up the multicast sockets, we need a destination address to listen
on or else we error.
2010-03-10 11:21:20 +01:00
Wim Taymans
574447b092 rtspsrc: handle ipv6 listening ports when needed
Add some code to make udpsrc listen on an ipv6 address when needed. The
detection of IPV6 is not yet implemented.
2010-03-10 11:21:20 +01:00
Wim Taymans
38f2b4735d rtspsrc: send keep alive when paused
When we are paused, send keep alive messages to the server so that our session
doesn't time out when we go back to playing later.
2010-03-10 11:21:18 +01:00
Wim Taymans
66709a7a68 rtspsrc: configure multicast correctly
Take the transport destination for multicast.
Disable loop and autojoin for multicast on the udpsinks.
2010-03-08 17:48:46 +01:00
Wim Taymans
a0b651bf5b rtspsrc: avoid stopping NULL tasks
Check the task for NULL, it could be paused and set to NULL before.
2010-02-16 19:54:32 +01:00
Mark Nauwelaerts
87e80aab57 rtspsrc: fix typo in debug message 2010-02-16 16:07:21 +01:00
Wim Taymans
8d814f3782 rtpbin: pass running_time to jitterbuffer pause
Pass the current running time to the jitterbuffer when pausing or resuming so
that it calculate the right offsets.
Small cleanups and comments.
Set the default rtspsrc latency to 2 seconds.
2010-02-12 17:22:54 +01:00
Wim Taymans
c2dfc94b1d rtspsrc: cleanup properties
Use more default constants.
Use static strings param flag.
Init properties explicitly instead of letting gobject do this.
2010-02-12 15:20:07 +01:00
Wim Taymans
c35a984801 rtspsrc: free transports on errors
See #608564
2010-02-01 19:32:11 +01:00
Wim Taymans
8c5a822250 rtspsrc: fix on-npt-stop signal warnings for RDT
The RDT manager does not implement this signal so we need to check for it before
trying to connect to it.
2010-01-05 12:23:16 +01:00
Wim Taymans
a65240d1c1 rtspsrc: fix some comments, remove property check
Fix some comments, clarify some FIXMEs
Remove the on-ntp-stop signal check now that the jitterbuffer is in
-good and we know that it supports this signal.
2009-12-24 22:23:01 +01:00
Thiago Santos
ac03ad782a rtspsrc: Parse all rtpinfo entries
Do not forget to parse all rtp-info entries, instead of
parsing the first one only.

Fixes #605222
2009-12-24 17:08:22 -03:00
Wim Taymans
b8c2ccce4e rtspsrc: handle NULL and empty transport strings
When an RTSP extension returns NULL or an empty transport string, just ignore it
and try to get the next possible transport. Fixes playback of RealMedia streams.
2009-12-10 18:45:55 +01:00
Wim Taymans
6a44d8e198 rtspsrc: install event function on internal RTCP pad
Install a custom event function on the internal RTCP pad so that we can reply
TRUE to a latency event.
2009-12-10 18:45:55 +01:00
Tim-Philipp Müller
24b93d82ec rtspsrc: fix major memory leak when playing back rtsp video streams
Don't forget to unref QoS, navigation and latency events when
dropping them.
2009-12-04 11:14:03 +00:00
Bastien Nocera
efc611e420 Add user-id and user-pw properties
So that one doesn't need to modify the URL to have access
to authenticated RTSP streams.

fixes #601728
2009-11-18 17:27:19 +01:00
Wim Taymans
6725c91387 rtsp: handle events in TCP mode
We need to handle events in TCP mode so that we can reply to the LATENCY event
with TRUE.
2009-10-15 13:20:26 +02:00
Wim Taymans
88884cfddb rtspsrc: forward events into the rtpbin
Only catch the SEEK event on the srcpad and let other events enter the rtpbin.
2009-10-14 17:01:51 +02:00
Mark Nauwelaerts
50d5c8dce5 rtspsrc: if transport protocol unsupported, try another one
Also change error message to more accurately reflect cases in which
it can occur.
2009-09-25 16:47:39 +02:00
Arnout Vandecappelle
19455200b1 rtspsrc: fix memory leak
In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth
header items and then passes them to gst_rtsp_connection_set_auth_param()
without freeing.

Fixes #594133
2009-09-08 13:30:29 +02:00
Wim Taymans
784b95ddbf rtspsrc: don't add non-utf8 chars to structures 2009-08-03 18:13:46 +02:00
Luc Deschenaux
f96e900a64 rtspsrc: put all SDP attributes on caps
Put the SDP attributes on the caps too so that they can be used by
depayloaders.

See #564437
2009-08-03 17:21:44 +02:00
Mark Nauwelaerts
a905ef233e rtspsrc: do not leak timeout message 2009-07-09 11:34:40 +02:00
Wim Taymans
81d7a297f7 rtspsrc: use same protocols after redirect
After a redirect we want to use the same protocols that we were using for the
current url.
2009-06-23 16:39:36 +02:00
Patrick Radizi
a95c049f76 rtspsrc: Add RTP blocksize functionality
Add property to make the client suggest a blocksize to the server.
Fixes #585549
2009-06-12 16:06:28 +02:00
Wim Taymans
b9ddf22340 rtspsrc: set the right state on rtpbin
We need to set the state of gstrtpbin to the same state as our source elements.
This fixes fallback to TCP again.
2009-06-04 15:19:05 +02:00
Patrick Radizi
301fc8a712 rtspsrc: fix memory leak of messages
Free messages correctly.
Fixes #577318
2009-05-25 10:57:59 +02:00
Wim Taymans
047618849a rtspsrc: make fakesrc silent
Make the fakesrc that is responsible for sending dummy packets silent.
2009-05-24 19:32:17 +02:00
Wim Taymans
5d3168e558 rtspsrc: don't send teardown before setup
Don't send a TEARDOWN request when we did not manage to successfully setup a
stream.
2009-05-24 16:33:42 +02:00
Wim Taymans
732704c007 rtspsrc: Fix find_stream_by_* functions
Fix various version of find_stream_by_* by not trying to convert an int to a
pointer and vice versa, for portability reasons.

Fixes #581333
2009-05-04 18:55:12 +02:00
Chris Winter
752cfb16fe rtspsrc: fix dummy nat packet logic
Fix a typo in the dummy NAT packet sending code.

Fixes #581329
2009-05-04 18:32:05 +02:00
Mark Nauwelaerts
959a9b494b rtspsrc: avoid errors after server eof
Server eof (e.g. connection closed) is announced as connection closed,
so better record state and act accordingly to prevent (read/write)
errors during subsequent teardown/cleanup sequences.  #Fixes 580851.(c).
2009-05-04 17:01:35 +02:00
Mark Nauwelaerts
734548a34f rtspsrc: also set base_time on src after flush
timestamps following flush/seek should be consistent between
UDP and TCP interleaved case.  Fixes #580851.(b).
2009-05-04 17:01:28 +02:00
Mark Nauwelaerts
20c7be5741 rtspsrc: sanity checks on range info
A max range that overflows should not be trusted,
nor should a max range that equals the min range.
Fixes #580851.(a).
2009-05-04 17:01:20 +02:00
Wim Taymans
56656dd03d rtspsrc: use SKIP flag to use SCALE headers
We can use the SKIP seek flag to instruct the server to send data faster then
normal but with the same bandwidth.
Fixes #537609
2009-05-04 16:18:23 +02:00
Wim Taymans
de0a2575fc rtspsrc: release state lock before stopping task
We need to release the state lock before trying to wait for the task to end
because the task might also take the lock.

Fixes #577671
2009-04-29 18:09:07 +02:00
Patrick Radizi
5b86c66e8a rtspsrc: fix some more pad leaks
Fix some pad leaks.
See #577318.
2009-04-22 15:27:24 +02:00
Edward Hervey
4c60f9ef29 rtspsrc: Remove dead assignment.
t is being overwritten after, before it's used.
2009-04-18 18:51:29 +02:00
Edward Hervey
45c6690e26 rtspsrc: Remove dead assignment. 'res' isn't read after. 2009-04-18 18:51:29 +02:00
Edward Hervey
817d7a30c3 rtspsrc: Remove unused variable. 'res' is never read. 2009-04-18 18:51:29 +02:00
Edward Hervey
08a090c89c rtspsrc: Remove dead variable. 'stream' is never read after. 2009-04-18 18:51:29 +02:00
Edward Hervey
0cb5b42d54 Remove trivial unused variables detected by CLang static analyzer. 2009-04-18 18:51:28 +02:00
Josep Torra
dfb375daa1 rtspsrc: mark discont on the streams as was said the debug line
After a seek mark all streams with discont as it was said in the debug line.
Fixes that buffers after a seek are generated without a valid timestamp.
2009-04-18 14:32:40 +02:00
Josep Torra
ec2d6053a0 rtspsrc: map GST_RTSP_EEOF to EOS on server requests
Permit properly handle the EOS condition when server report it in a request.
2009-04-18 08:50:46 +02:00
Wim Taymans
b6bf3ba7d3 rtspsrc: allow http:// on the proxy setting
Allow and ignore http:// at the start of the proxy setting, like
souphttpsrc.
Fixes #573173
2009-04-02 22:41:01 +02:00
Wim Taymans
40f6ed8875 rtspsrc: don't leak the udpsrc pad
Fix memory leak in rtspsrc because we didn't unref the udpsrc pad.
See #577318
2009-04-02 21:08:48 +02:00
Tim-Philipp Müller
cb15d09c4a rtspsrc: don't emit ugly warnings with older rtpjitterbuffer versions
The on-npt-stop signals was added only recently to rtpjitterbuffer in
-bad, so check if the signal exists before g_signal_connect()ing to
it, to avoid warnings.
2009-04-01 12:29:33 +01:00
Wim Taymans
b037369d5b rtspsrc: add proxy support 2009-03-31 19:08:37 +02:00
Wim Taymans
fd18185d44 rtspsrc: link to the on_npt_stop signal to EOS
Connect to the on_npt_stop signal of the session manager to schedule the EOS
actions.
2009-03-27 17:49:15 +01:00
Tim-Philipp Müller
37634c2afb rtspsrc: better error message when the RTSP extension for Real streams is missing
Try to post a decent error message when it looks like we're failing
because the Real RTSP extension plugin is missing. Also add i18n
bits for rtspsrc so our error messages get translated.
2009-03-25 17:54:35 +00:00
Wim Taymans
8cf0e9ff87 rtspsrc: add some debug for the timestamps
When timestamping in TCP mode, log the first timestamp we put on the buffers.
2009-03-16 19:17:24 +01:00
Wim Taymans
7782c9f890 rtspsrc: don't send PAUSE when not connected
don't send a PAUSE request when we are no longer connected.
2009-03-12 20:39:35 +01:00
Wim Taymans
515d623dcc rtspsrc: fix timeout check
---
2009-03-11 18:00:02 +01:00
Wim Taymans
636cd65ebf rtspsrc: fix range parsing
Fix parsing of the range headers.
2009-03-05 14:09:03 +01:00
Wim Taymans
5a5ba49c9b rtspsrc: fix memory leak in close
Close the connection even when we fail to send the teardown message.
Use the connection url (which is a copy of the src url).
2009-03-04 16:31:57 +01:00
Wim Taymans
dfb2d1b7d7 rtspsrc: fix do-rtcp property description
---
2009-03-04 12:29:50 +01:00
Wim Taymans
81f25317e6 rtspsrc: add support for http tunneling
Add support for http tunneling and a new rtsph:// uri for it.
See #573173.
2009-03-02 16:09:23 +01:00
Patrick Radizi
c7dd6a4902 rtspsrc: add property to disable RTCP
Some old servers don't like us doing RTCP and thus we need a property to disable
it. See #573173.
2009-02-26 19:03:52 +01:00
Mark Nauwelaerts
21cb00aa9c rtspsrc: perform UDP SETUP according to MS RTSP spec
MS RTSP spec states that the UDP port pair used in subsequent SETUP
requests for various streams must be identical (since there will actually
be only 1 stream of muxed asf packets).  Following traditional specs and
using different port pairs in the SETUPs for separate streams will result
in all but the first one failing and only one stream being streamed.

So, in appropriate circumstances, retry UDP SETUP using previously used
port pair.  Fixes #552650.
2009-02-23 22:47:55 +01:00
Wim Taymans
a08d75b892 Call new receive_request method
Call the receive_request extension methods so that extensions can handle the
server request if they want.
2009-02-23 11:42:53 +01:00
Wim Taymans
1dc5c34143 rtspsrc: Keep track of connected state
Keep track of the state of the connection and don't try to send TEARDOWN when
the server has closed the connection.
2009-02-04 11:38:30 +01:00
Stefan Kost
a99d3f8769 Update and add documentation for plugins with no deps (gst).
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
2009-01-28 12:32:59 +02:00
Wim Taymans
16799b6b16 Free leftover udp ports (if any) when a setup request fails. 2009-01-22 12:21:29 +01:00
이문형
42f6a2bca1 gst/rtsp/gstrtspsrc.c: Prevent further read/write actions taken to the connect-failed socket by erroring out quickly....
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Prevent further read/write actions taken to the connect-failed socket by
erroring out quickly. See #562258.
2008-11-27 11:22:56 +00:00
Wim Taymans
0b5fea8568 gst/rtsp/gstrtspsrc.c: Add some more debugging.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (new_session_pad),
(gst_rtspsrc_parse_range):
Add some more debugging.
Use the reanges received from the server unconditionally.
Fixes #561625.
2008-11-24 12:20:29 +00:00
Eric Zhang
be3906c918 gst/rtsp/gstrtspsrc.*: Add property to configure NAT traversal method.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes #559545.
2008-11-13 16:11:16 +00:00
Wim Taymans
21edbcc566 gst/rtsp/gstrtspsrc.c: Only send one transport at a time for improved compatibility with some broken servers. See #53...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
(gst_rtspsrc_change_state):
Only send one transport at a time for improved compatibility with some
broken servers. See #537832.
2008-11-11 16:00:48 +00:00
Wim Taymans
8a2bcfecb0 gst/rtsp/gstrtspsrc.c: Only pause/play in the seek handler when the source was playing.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_perform_seek):
Only pause/play in the seek handler when the source was playing.
Fixes #529379.
2008-11-11 15:16:31 +00:00
Eric Zhang
499c3e520e gst/rtsp/gstrtspsrc.c: Pause the RTSP stream before doing a new play request.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek),
(gst_rtspsrc_stream_configure_udp_sink):
Pause the RTSP stream before doing a new play request.
Make sure that adding the udpsinks does not cause the rtspsrc to become
a sink. Fixes #559547.
2008-11-10 12:13:21 +00:00
Wim Taymans
539627e049 gst/rtsp/gstrtspsrc.c: Return TRUE instead of FALSE from the event handler when we swallowed the event.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event):
Return TRUE instead of FALSE from the event handler when we swallowed the
event.
2008-10-09 14:27:12 +00:00
Wim Taymans
b1dfdc758e gst/rtsp/gstrtspsrc.c: Don't assume the server supports PAUSE by default. Fixes #551048.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
Don't assume the server supports PAUSE by default. Fixes #551048.
2008-09-25 12:07:46 +00:00
Wim Taymans
bf8777356b gst/rtsp/gstrtspsrc.c: Handle the case where we cannot do desribe or when the describe result does not contain a vali...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
Handle the case where we cannot do desribe or when the describe result
does not contain a valid SDP message.
2008-09-23 18:08:56 +00:00
Wim Taymans
dd54e000ea gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes #546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
2008-08-20 17:30:19 +00:00
Wim Taymans
0dfa54f450 gst/rtsp/gstrtspsrc.c: Don't try to configure RTCP back to the server when the server did not give us a valid port nu...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Don't try to configure RTCP back to the server when the server did not
give us a valid port number.
2008-08-20 11:48:46 +00:00
Aurelien Grimaud
1e64691186 gst/rtsp/gstrtspsrc.c: Improve udp port setup. Fixes #545710.
Original commit message from CVS:
Patch by: Aurelien Grimaud <gstelzz at yahoo dot fr>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_alloc_udp_ports):
Improve udp port setup. Fixes #545710.
2008-08-05 13:57:57 +00:00
Stefan Kost
9f886ee1f2 gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi().
Original commit message from CVS:
* gst/rtp/gstrtpvrawdepay.c:
Include stdlib.h for atoi().
* gst/rtsp/gstrtspsrc.c:
Use floating point math for latencies < 0 sec in log output.
2008-07-07 10:30:51 +00:00
Wim Taymans
8d901b4bfc gst/rtsp/gstrtspsrc.c: Set udpsrc for receiving data from multicast groups to PAUSED instead of leaving them in READY...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_mcast):
Set udpsrc for receiving data from multicast groups to PAUSED instead of
leaving them in READY. Fixes #537832.
2008-06-12 17:30:06 +00:00
Peter Kjellerstedt
d60878ab14 gst/rtsp/gstrtspsrc.c: Use the new gst_rtsp_connection_get_ip() to access the IP address of a GstRTSPConnection since...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Use the new gst_rtsp_connection_get_ip() to access the IP address
of a GstRTSPConnection since it is a private member.
2008-06-04 11:59:18 +00:00
Wim Taymans
487b784b4f Don't use gst_element_get_pad(), it's a bad method.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset),
(do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset),
(do_toggle_element):
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws),
(gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas),
(gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr):
* tests/icles/videocrop-test.c: (test_with_caps),
(video_crop_get_test_caps):
Don't use gst_element_get_pad(), it's a bad method.
2008-05-21 17:39:38 +00:00
Wouter Cloetens
5506fbfc48 gst/rtsp/gstrtspsrc.c: Support Digest authentication. Fixes #532065.
Original commit message from CVS:
Based on patch by: Wouter Cloetens  <wouter at mind be>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws),
(gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item),
(gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr),
(gst_rtspsrc_setup_auth):
Support Digest authentication. Fixes #532065.
2008-05-08 16:58:02 +00:00
Sjoerd Simons
89b114fe44 gst/rtsp/gstrtspsrc.c: Don't leak file descriptors on error. Fixes #531532.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_open):
Don't leak file descriptors on error. Fixes #531532.
2008-05-05 11:19:13 +00:00
Wim Taymans
f9646f3722 gst/rtsp/gstrtspsrc.c: Ref caps as the return value for the request_pt_map signal.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (request_pt_map),
(gst_rtspsrc_configure_caps):
Ref caps as the return value for the request_pt_map signal.
Remove some caps weirdness when configuring a stream. See #528245.
2008-04-21 08:21:14 +00:00
Ole André Vadla Ravnås
110a0ea563 gst/rtsp/gstrtspsrc.c: Call WSAStartup() and WSACleanup before using the Winsock API.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize):
Call WSAStartup() and WSACleanup before using the Winsock API.
See #520808.
2008-03-17 15:56:01 +00:00
Wim Taymans
7f0745bb7f gst/rtsp/gstrtspsrc.c: Post the server response code in an error message instead of a generic 'error' message. Fixes ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Post the server response code in an error message instead of a generic
'error' message. Fixes #517237.
2008-02-22 09:56:03 +00:00
Wim Taymans
a1abaa3bfe gst/rtsp/gstrtspsrc.c: Init values to -1 instead of the default 0 value.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream):
Init values to -1 instead of the default 0 value.
Fixes #516524.
2008-02-18 11:13:35 +00:00
Sébastien Moutte
f0690e19ea gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is defined
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c:
Include unistd.h only if HAVE_UNISTD_H is defined
* win32/common/config.h.in:
* win32/common/config.h:
Define socklen_t as it seems it's not defined in default
Visual Studio headers.
* win32/vs6/libgstalpha.dsp:
* win32/vs6/libgstapetag.dsp:
* win32/vs6/libgstavi.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstvideomixer.dsp:
Update project file dependencies and add new source files
2008-02-07 19:13:56 +00:00
Tim-Philipp Müller
7c7b58e839 gst/rtsp/gstrtspsrc.c: Use g_ascii_strtoll() instead of atoll, which is only available in C99.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use g_ascii_strtoll() instead of atoll, which is only
available in C99.
2008-01-28 12:17:02 +00:00
Wim Taymans
8a72bf80e7 As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
Original commit message from CVS:
As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo):
Use atoll to parse the rtptime with enough precision. Fixes #509329.
2008-01-14 12:35:23 +00:00
Tim-Philipp Müller
11118eabb9 gst/: Initialise variables to work around (false) 'foo might be used uninitialized in this function' warnings by gcc-...
Original commit message from CVS:
* gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
Initialise variables to work around (false) 'foo might be used
uninitialized in this function' warnings by gcc-3.3.3 (#509298).
2008-01-14 12:11:43 +00:00