Commit graph

73 commits

Author SHA1 Message Date
Tim-Philipp Müller
87aa29d2cf rtspsrc: make connection-speed property a guint64 2011-11-24 01:19:32 +00:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +02:00
Marc Leeman
ce276d903c rtspsrc: allow sending short RTSP requests to a server
Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
GStreamer, but do accept the short header as sent by Live555.

This patch makes the extending the request optional by adding a property
(short-header).

Fixes #655805.

API: GstRTSPSrc:short-header
2011-09-05 13:26:06 +02:00
Wim Taymans
c731cd3d95 rtsp: port to 0.11 2011-06-09 17:52:34 +02:00
Mark Nauwelaerts
f7ddf811d7 rtspsrc: fix and improve async handling
Simplify the command handling; passing a command to thread means we really
want it to get the message, which means to always flush provided the command
can handle being interrupted.  Command thread indicates whether command
allows interruption and ensure non-flushing connection as it subsequently
needs it.

In particular, this also makes the TEARDOWN sequence interruptable
and also prevents races where _loop_ could miss a command and would
continue receiving (or at least trying to).

See #632504.
2011-05-17 11:56:22 +02:00
Wim Taymans
ddfcd8bbfd rtspsrc: open on play and pause when not done yet
With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
2011-05-17 11:55:34 +02:00
Wim Taymans
220e47adcf rtspsrc: implement more async handling
Remove some old locks.
Make sure we never go into the loop function when flushing.
2011-05-17 11:55:20 +02:00
Wim Taymans
dae679e560 rtspsrc: small header cleanups 2011-05-17 11:55:15 +02:00
Mark Nauwelaerts
2738917852 rtspsrc: improve recovery from failed seek
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process.  So, query position prior
to seek and use upon failed seek.
2011-03-09 17:18:09 +01:00
Wim Taymans
12bc7258b9 rtspsrc: improve RTP session handling
Store the RTP session in the stream so that we can more efficiently
perform actions on the stream based on RTP signals.
2010-12-23 15:24:29 +01:00
Mark Nauwelaerts
c25625c31c rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response 2010-12-10 12:09:49 +01:00
Wim Taymans
7ed250c793 rtspsrc: select multicast transports in a smarter way
When we see a multicast address in the SDP connection, only try to negotiate a
multicast transport with the server.

Fixes #634093
2010-12-02 19:16:47 +01:00
Wim Taymans
ae84ae1b36 rtspsrc: add rtsp-sdp protocol support
Allow setting an SDP with the rtsp-sdp:// url.

Based on patch from Marco Ballesio.

See #628214
2010-09-10 12:14:21 +02:00
American Dynamics
5999e8e716 rtspsrc: Add property to configure udpsrc buffer size
Add a new udp-buffer-size property to configure the buffer-size on the udpsrc
elements.

Fixes #628058
2010-09-06 12:22:11 +02:00
Arnaud Vrac
c6f47c34fb rtspsrc: add port-range property to rtspsrc
To support setups with firewall/ipsec, it is useful for an rtsp client to be
able to set the range of ports that can be used for rtp/rtcp reception.
Allows this by adding a "port-range" property to the rtspsrc element.

Fixes #625153
2010-07-26 17:47:35 +02:00
Wim Taymans
966ced2208 rtspsrc: factor out the connections
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
2010-06-18 15:13:06 +02:00
Wim Taymans
e6ec5cce2e rtspsrc: respect aggregate control attributes
when the SDP specifies an aggregate control url, use that for playback
control.

Fixes #619531
2010-06-14 19:24:14 +02:00
Wim Taymans
db3c4e7f46 rtspsrc: fall back to SDP ports instead of server_port
In multicast, fall back to the ports in the SDP instead of the server_port
attribute as this is more in line with the RFC.
2010-05-07 12:51:05 +02:00
Wim Taymans
ef4d2901aa rtspsrc: use the SDP connection info in multicast
Parse the connection info from the SDP.
When we need to configure the multicast destination, fall back to the SDP
connection info when the transport did not specify a destination and ttl.

Fixes #617537
2010-05-06 16:52:26 +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
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
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
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
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
b037369d5b rtspsrc: add proxy support 2009-03-31 19:08:37 +02:00
Wim Taymans
636cd65ebf rtspsrc: fix range parsing
Fix parsing of the range headers.
2009-03-05 14:09:03 +01:00
Patrick Radizi
51200cad41 rtspsrc: add the .h file change too
Add the .h file change for the new property.
2009-02-26 19:05:06 +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
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
eb5e87944c gst/rtsp/gstrtspsrc.*: Implement redirect for the DESCRIBE reply. Fixes #506025.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Implement redirect for the DESCRIBE reply. Fixes #506025.
2007-12-31 13:27:32 +00:00
Wim Taymans
5274c3f4e2 gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
* gst/rtsp/gstrtspsrc.h:
Parse bandwidth modifiers, they are not yet configured in the session
manager because we don't have an API for that yet.
2007-10-01 16:34:56 +00:00
Wim Taymans
bea9010658 gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
In TCP mode, only timestamp the first buffer. TCP is not real time and
it does not make sense to try to skew compensate, also some servers send
the first batch of data in a burst.
2007-09-28 14:56:19 +00:00
Wim Taymans
4683ff80d3 gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
Set timestamps on RTP buffers in interleaved mode.
Mark first buffers with a DISCONT.
Remove flush hack now that sync for live sources has been figured out.
2007-09-26 20:12:52 +00:00
Wim Taymans
5592bdd459 gst/rtsp/gstrtspsrc.*: Fix method detection again.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
2007-08-22 15:01:29 +00:00
Wim Taymans
0dcafb0635 gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes #455808.
2007-08-17 17:08:11 +00:00
Wim Taymans
6ef7055041 gst/rtsp/gstrtspsrc.*: Improve timeout handling.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
2007-08-17 14:15:19 +00:00
Wim Taymans
41f0496738 gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
2007-08-16 11:47:19 +00:00
Wim Taymans
fa9c47f14d gst/rtsp/: Use shiny new RTSP and SDP library.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/base64.c:
* gst/rtsp/base64.h:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
(gst_rtsp_ext_list_setup_media),
(gst_rtsp_ext_list_configure_stream),
(gst_rtsp_ext_list_get_transports),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c:
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c:
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtspmessage.c:
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsprange.c:
* gst/rtsp/rtsprange.h:
* gst/rtsp/rtsptransport.c:
* gst/rtsp/rtsptransport.h:
* gst/rtsp/rtspurl.c:
* gst/rtsp/rtspurl.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.c:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/test.c:
Use shiny new RTSP and SDP library.
Implement RTSP extensions using the new interface.
Remove a lot of old code.
2007-07-25 18:50:08 +00:00
Wim Taymans
cf20f497cc gst/rtsp/gstrtspsrc.*: For container formats we only need to activate one of the streams so that we correctly signal ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (gst_rtspsrc_setup_streams):
* gst/rtsp/gstrtspsrc.h:
For container formats we only need to activate one of the streams so
that we correctly signal no-more-pads. Fixes #451015.
2007-06-27 11:04:47 +00:00
Wim Taymans
e04f7a828f gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Add TCP timeout property and use it for all TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_write), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
Make connect and writes cancelable and make them use the timeout.
2007-05-18 11:39:12 +00:00
Peter Kjellerstedt
7ef62aac45 gst/rtsp/: Make channel guint8 where possible.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
(rtsp_message_get_header):
* gst/rtsp/rtspmessage.h:
Make channel guint8 where possible.
Make rtsp_message_init_data() take the channel as a guint8.
* gst/rtsp/rtspdefs.c:
Fixed a typo: Timout -> Timeout
* gst/rtsp/rtspdefs.h:
Make RTSP_CHECK() behave as a statement.
* gst/rtsp/sdpmessage.c:
Avoid a compiler warning in INIT_ARRAY().
Fixes #437692.
2007-05-12 16:37:50 +00:00
Wim Taymans
02fa0a7992 gst/rtsp/: Preliminary seek support.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_handle_src_event),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.h:
Preliminary seek support.
Activate internal pads so that we can receive events on them.
Don't try to parse a range string when it's NULL.
2007-05-11 15:09:39 +00:00
Wim Taymans
24e51b3c73 gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
2007-05-02 19:32:58 +00:00
Wim Taymans
92396be152 gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
Wim Taymans
530f214bd5 gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect state changes with a lock.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(parse_line):
* gst/rtsp/rtspconnection.h:
Remove some unused stuff.
2007-04-26 10:08:27 +00:00
Wim Taymans
6937be1a09 gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property):
* gst/rtsp/gstrtpdec.h:
Add dummy latency property to be backwards compat with rtpbin.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Add latency property and configure in the session manager.
Don't set invalid clock-base and seqnum-base on caps, some servers
sometimes don't send them.
2007-04-25 15:55:32 +00:00
Wim Taymans
1beeda3ff2 gst/rtsp/gstrtspsrc.*: Parse server address from SDP.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_stream_free), (request_pt_map),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Parse server address from SDP.
Hook up a udpsink to send RTCP back to the server.
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/rtsptransport.h:
Add some docs.
2007-04-25 08:36:46 +00:00
Wim Taymans
86a4c1c6b0 gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
* gst/rtsp/gstrtpdec.h:
Make backward compat with rtpbin by adding the request-pt-map signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams):
* gst/rtsp/gstrtspsrc.h:
Implement request-pt-map signals instead of setting caps on the buffers
for the session manager.
2007-04-12 08:21:28 +00:00
Wim Taymans
f80444aaec gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
Wim Taymans
8f5fb88b5a gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Handle default clock-rates for static payload types, rearrange stuff so
that the rtpmap field in the sdp can override the defaults.
Parse RTP-Info field to get the seqnum and timebase fields that should
go in the caps.
Delay configuring caps after we got the RTP-Info from the PLAY reply from
the server.
2007-03-25 15:34:42 +00:00