| Commit message (Collapse) | Author | Age | Files | Lines |
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Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.
Fixes #594490
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fixes #594254
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Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
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There is no need to provide a clock.
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Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the
application that the NPT stop position has been reached.
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needed and pass new info in the method call.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_client),
(gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream),
(gst_rtp_bin_class_init), (new_ssrc_pad_found):
* gst/rtpmanager/gstrtpbin.h:
Remove a lot of per stream state that is not needed and pass new info in
the method call.
Add signal to reset sync parameters.
Avoid parsing the caps to get a clock_base, we get this from the sync
signal now.
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Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
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becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose new jitterbuffer property in rtpbin too.
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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.
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the session managers with them.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name),
(gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose SDES items as properties and configure the session managers with
them.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_set_property):
Fix SSRC property.
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Fixes #478566.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_active),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_active), (rtp_session_process_rb):
* gst/rtpmanager/rtpsession.h:
Add notification of active SSRCs to various RTP elements. Fixes #478566.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(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_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.
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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.
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jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
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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.
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Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_set_flushing_unlocked):
Fix leak when flushing.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add clear-pt-map signal.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
Init clock-rate to -1 to mark unknow clock rate.
Fix flushing.
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Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
(create_recv_rtp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_finalize),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_request_new_pad):
Protect lists and structures with locks.
Return FLOW_OK from RTCP messages for now.
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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.
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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.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
* gst/rtpmanager/gstrtpbin.h:
Provide a clock.
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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.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
Some more ghostpad magic.
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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.
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