Commit graph

42 commits

Author SHA1 Message Date
Paul HENRYS
0e91c949d8 rtpptdemux: forward sticky events and then set caps
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
2013-02-07 14:38:20 +01:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Wim Taymans
5cce960baa ptdemux: set caps after activating the pad
Set the caps after we activated the pad or else it will just fail.
2012-03-07 15:02:44 +01:00
Olivier Crête
33a6d1921f rtpptdemux: Forward sticky events 2012-01-27 19:05:24 +01:00
Olivier Crête
7b1f8cb8f0 rtpptdemux: Protect all uses pad list with OBJECT LOCK
Actually protect the entire pad list and use it in a thread safe
way.
2012-01-27 19:05:24 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
2012-01-25 12:49:11 +01:00
Tim-Philipp Müller
f10e8192fa rtpptdemux: plug pad leak in error code path
Based on patch by: Stig Sandnes <stig.sandnes@cisco.com>

Don't leak srcpad if there are no caps.

https://bugzilla.gnome.org/show_bug.cgi?id=667820
2012-01-13 11:02:24 +00:00
Edward Hervey
86a57e3546 rtpmanager: Initialize GstRTPBuffer before usage 2011-12-05 18:40:12 +01:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
a95acb7122 make %u in all request pad templates 2011-11-04 11:58:22 +01:00
Mark Nauwelaerts
ef02634dc6 rtpmanager: port to 0.11
* use G_DEFINE_TYPE
* adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces
* misc caps and segment handling changes

FIXME: also relies on being able to pass caps along with a buffer,
which has no evident equivalent yet, so that either needs one,
or still needs quite some code path modification to drag along caps.
2011-07-06 10:16:12 +02:00
Sebastian Dröge
31af4fe33e rtp: Unref events if the parent element disappeared 2011-04-08 15:20:51 +02:00
Ole André Vadla Ravnås
046f170d6a rtpmanager: fix pad callbacks so they handle when parent goes away
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.

This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
2011-04-08 15:16:56 +02:00
Olivier Crête
589b254ce5 rtpptdemux: Tag upstream custom events with payload type 2011-02-01 16:50:25 +01:00
Thijs Vermeir
244a35a226 rtpptdemux: fix memleak on custom downstream events
by not sending custom downstream event twice and fix memleak when
not handling the event

https://bugzilla.gnome.org/show_bug.cgi?id=623196
2010-06-30 12:39:09 +02:00
Sebastian Dröge
f16ed4a91c gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 17:52:40 +02:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Håvard Graff
58b9de4cca rtpptdemux: only forward the lost-event to the last seen pt-number
forward all events on all pads except for the PacketLost event, which we want to
forward to the last seen pt pad.

Fixes #598377
2009-10-14 12:28:55 +02:00
Stefan Kost
0a7ef67ad0 docs: fix gtk-doc warnings 2009-09-10 10:28:48 +03:00
Wim Taymans
5ab3e10594 gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
Do not try to keep track of the clock-rate ourselves but simply get the
value from the jitterbuffer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add some debug info.
Pass the clock-rate to the jitterbuffer.
Also pass the clock-rate along with the rtp timestamp when getting the
sync parameters.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix some debug.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of clock-rate changes and return the clock-rate together with
the rtp timestamps used for sync.
Don't try to construct timestamps when we have no base_time.
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Request a new clock-rate when the payload type changes.
Reset the jitter calculation when the clock-rate changes.
2009-08-11 02:30:38 +01:00
Stefan Kost
138c2b7cf9 gst/: More doc updates. More xrefs.
Original commit message from CVS:
* gst/deinterlace/gstdeinterlace.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/sdp/gstsdpdemux.c:
More doc updates. More xrefs.
2009-08-11 02:30:36 +01:00
Stefan Kost
2d1ccbf52e Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2009-08-11 02:30:36 +01:00
Olivier Crete
3c5cf0cd38 gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(new_ssrc_pad_found):
Ref caps when inserting into the cache.
Don't leak pads.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_query):
Avoid a caps leak.
Don't leak refcount in query.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_chain):
Avoid caps leaks.
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(gst_rtp_session_init), (return_true),
(gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps),
(gst_rtp_session_clock_rate):
Ref caps when inserting into the cache.
Fix some more caps leaks. Fixes #528245.
2009-08-11 02:30:34 +01:00
Wim Taymans
4cc70a0c22 gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client),
(gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Don't leak a padname.
Don't leak client streams list.
Lock rtpbin when associating streams. Fixes #528245.
2009-08-11 02:30:34 +01:00
Wim Taymans
a965ebff09 gst/rtpmanager/: Clean up the dynamic pads when going to READY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
(gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
(gst_rtp_ssrc_demux_change_state):
Clean up the dynamic pads when going to READY.
2009-08-11 02:30:32 +01:00
Wim Taymans
49e501a647 gst/rtpmanager/: Add signal to notify of an SDES change.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_sdes), (rtp_session_process_sdes):
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.c:
* gst/rtpmanager/rtpstats.h:
Add signal to notify of an SDES change.
Fix object type in the signal callbacks.
2009-08-11 02:30:32 +01:00
Wim Taymans
4556ccb666 gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map),
(gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init):
Fix caps refcounting for payload maps.
When clearing payload maps, also clear sessions and streams payload
maps.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain),
(find_pad_for_pt):
Implement clearing the payload map.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink):
Forward flush events instead of leaking them.
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_rtcp_sink_event):
Correctly refcount events before pushing them.
2009-08-11 02:30:31 +01:00
Wim Taymans
a93348cc6d gst/rtpmanager/: Various leak fixes.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (free_session),
(get_client), (free_client), (gst_rtp_bin_associate),
(free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_finalize):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize),
(gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose):
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
* gst/rtpmanager/rtpsession.h:
Various leak fixes.
2009-08-11 02:30:30 +01:00
Tim-Philipp Müller
10d6ba4d61 Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF...
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
registers a GType that's different than the GstRTPFoo types that
farsight registers (luckily GType names are case sensitive). Should
finally fix #430664.
2009-08-11 02:30:29 +01:00
Wim Taymans
0c4fe985b6 Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
(gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpclient.c: (create_stream),
(gst_rtp_client_request_new_pad):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpssrcdemux.c:
Rename elements to avoid conflict with farsight elements with the same
name. Fixes #430664.
2009-08-11 02:30:28 +01:00
Wim Taymans
2a8cfc6410 Document stuff.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_clear_pt_map):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_clear_pt_map):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Document stuff.
Add clear-pt-map action signal where needed.
2009-08-11 02:30:27 +01:00
Wim Taymans
3bc059707d gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
We always use fixed caps.
2009-08-11 02:30:27 +01:00
Wim Taymans
600afaaff9 gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
2009-08-11 02:30:26 +01:00
Wim Taymans
8bbea77a41 gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
Emit pt map requests and cache results.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps),
(gst_jitter_buffer_sink_setcaps),
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Emit request-pt-map signals.
2009-08-11 02:30:25 +01:00
Wim Taymans
03bf43d50e gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
Some more custom marshallers.
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
(pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
* gst/rtpmanager/gstrtpbin.h:
Prepare for caching pt maps.
Connect to signals to collect pt maps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add request_clock_rate signal.
Use scale insteat of scale_int because the later does not deal with
negative numbers.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_chain):
* gst/rtpmanager/gstrtpptdemux.h:
Implement request-pt-map signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
8c67b5d7dd gst/rtpmanager/: Added custom marshallers for signals.
Original commit message from CVS:
* gst/rtpmanager/.cvsignore:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpbin-marshal.list:
Added custom marshallers for signals.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Prepare for emiting pt map signals.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Fix signals.
2009-08-11 02:30:24 +01:00
Wim Taymans
9bfc641f0d gst/rtpmanager/gstrtpbin.*: Add debugging category.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
f0d1ab1c1f Add RTP session management elements. Still in progress.
Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.
2009-08-11 02:30:23 +01:00