mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-22 08:17:01 +00:00
Code Issues 1 Projects Releases Wiki Activity
7507 commits 44 branches 1510 tags 509 MiB
Commit graph

194 commits

Author SHA1 Message Date
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