Commit graph

52 commits

Author SHA1 Message Date
Wim Taymans
126e0fd0f4 gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
Install event handler on the rtcp_src pad, make LATENCY event return
TRUE.
2008-10-16 09:51:28 +00:00
Håvard Graff
b398d611a4 gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
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.
2008-10-07 18:54:41 +00:00
Wim Taymans
dbc5475c97 gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
(free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
(remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
(gst_rtp_bin_release_pad):
Release pads of the session manager.
Start implementing releasing pads of gstrtpbin.
* gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
(remove_recv_rtcp_sink), (remove_send_rtp_sink),
(remove_send_rtcp_src), (gst_rtp_session_release_pad):
Implement releasing pads in gstrtpsession.
2008-10-07 11:33:10 +00:00
Wim Taymans
a35d1dde42 gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender 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.
2008-09-05 13:52:34 +00:00
Wim Taymans
53025584c3 gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes #549409.
2008-08-28 15:21:45 +00:00
Wim Taymans
85b99b9077 gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
2008-08-13 14:31:02 +00:00
Peter Kjellerstedt
e6d85e6a1e gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_send_rtp):
* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp):
Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
pipeline is running normally.
2008-07-03 14:31:10 +00:00
Peter Kjellerstedt
56988f51e1 gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time().
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_send_bye_locked), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
(is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Do not mix the use of g_get_current_time() with gst_clock_get_time().
2008-07-03 13:47:19 +00:00
Stefan Kost
68993006c3 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.
2008-06-16 07:03:58 +00:00
Stefan Kost
15b9246617 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.
2008-06-12 14:49:18 +00:00
Peter Kjellerstedt
11ebf3a6aa gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
Make sure to unref the rtpsession returned by gst_pad_get_parent() to
prevent a memory leak.
2008-05-12 14:28:09 +00:00
Sjoerd Simons
08b3343e6b gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink):
Send RTP BYE command on EOS. Fixes bug #531955.
2008-05-08 06:23:39 +00:00
Wim Taymans
5fe12accc9 gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
(create_send_rtcp_src):
Set up some internal links functions for the RTCP and sync pads because
the defaults are really not correct.
Implement a query handler for the RTCP src pad, mostly to correctly
report about the latency.
2008-04-25 08:21:06 +00:00
Olivier Crete
4be1b4a30a 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.
2008-04-21 08:26:37 +00:00
Wim Taymans
0fcc94b64e 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.
2008-04-17 07:31:44 +00:00
Peter Kjellerstedt
fbc2a12aa5 gst/rtpmanager/: Avoid leaking pads in the RTP manager.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_session):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
Avoid leaking pads in the RTP manager.
2008-04-09 22:27:50 +00:00
Ole André Vadla Ravnås
0b94354a0d gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...
Original commit message from CVS:
Based on patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
(join_rtcp_thread), (gst_rtp_session_change_state):
Avoid a deadlock when joining the RTCP thread in PAUSED because it might
be blocked downstream. Also avoid spawning multiple rtcp threads.
Fixes #520894.
2008-03-11 11:36:03 +00:00
Wim Taymans
59d2dce4fd gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to parse the clock-rate instead of returning...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
If we find the caps in the cache, use it to parse the clock-rate instead
of returning an error. Fixes a TODO as found by Youness Alaoui.
2008-01-11 17:02:30 +00:00
Youness Alaoui
2e0d1efb0e gst/rtpmanager/: Make it possible to use different user_data for each of the callbacks.
Original commit message from CVS:
Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(rtp_session_set_process_rtp_callback),
(rtp_session_set_send_rtp_callback),
(rtp_session_set_send_rtcp_callback),
(rtp_session_set_sync_rtcp_callback),
(rtp_session_set_clock_rate_callback),
(rtp_session_set_reconsider_callback), (source_push_rtp),
(source_clock_rate), (rtp_session_process_bye),
(rtp_session_process_rtcp), (rtp_session_send_bye),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible to use different user_data for each of the callbacks.
Fixes #508587.
2008-01-11 16:45:57 +00:00
Wim Taymans
8fc6e320d0 gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do everything the upsteam peer pad can renegot...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
Don't set fixed caps, we can basically do everything the upsteam peer
pad can renegotiate to. Fixes #507940.
2008-01-09 14:39:44 +00:00
Wim Taymans
8b973428f3 gst/rtpmanager/: Post a message when the SDES infor changes for a source.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(on_ssrc_sdes):
Post a message when the SDES infor changes for a source.
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
Update some comments.
2007-12-10 18:36:04 +00:00
Wim Taymans
b99d637ab7 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.
2007-12-10 15:34:19 +00:00
Wim Taymans
582f643ee4 gst/rtpmanager/: Update comment.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
* gst/rtpmanager/rtpjitterbuffer.c:
Update comment.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_set_property), (gst_rtp_session_get_property):
Define some GObject properties to set SDES and other configuration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_ssrc_sdes), (rtp_session_set_bandwidth),
(rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
(rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
(rtp_session_get_sdes_string), (obtain_source),
(rtp_session_get_internal_source), (rtp_session_process_sdes),
(rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
(is_rtcp_time):
* gst/rtpmanager/rtpsession.h:
Add signal when new SDES infor has been found for a source.
Create properties for SDES and other info.
Simplify the SDES API.
Add method for getting the internal source object of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_finalize), (rtp_source_set_property),
(rtp_source_get_property), (rtp_source_set_callbacks),
(rtp_source_get_ssrc), (rtp_source_set_as_csrc),
(rtp_source_is_as_csrc), (rtp_source_is_active),
(rtp_source_is_validated), (rtp_source_is_sender),
(rtp_source_received_bye), (rtp_source_get_bye_reason),
(rtp_source_set_sdes), (rtp_source_set_sdes_string),
(rtp_source_get_sdes), (rtp_source_get_sdes_string),
(rtp_source_get_new_sr), (rtp_source_get_new_rb):
* gst/rtpmanager/rtpsource.h:
Add GObject properties for various things.
Don't leak the bye reason.
2007-12-10 11:08:11 +00:00
Ole André Vadla Ravnås
c67ad65c91 gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c:
Fix bad function signatures (#492798).
2007-11-02 21:45:38 +00:00
Jan Schmidt
1fe4050c0a gst/: Fix compiler warnings shown by Forte.
Original commit message from CVS:
* gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc):
* gst/librfb/rfbbuffer.h:
* gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer):
* gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain):
* gst/nsf/nes6502.c: (nes6502_execute):
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
* gst/real/gstrealvideodec.c: (open_library):
* gst/real/gstrealvideodec.h:
* gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink):
Fix compiler warnings shown by Forte.
2007-10-08 17:46:45 +00:00
Wim Taymans
4865b3bf14 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.
2007-10-08 10:39:35 +00:00
Wim Taymans
6e5d23b3d7 gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead of popping it off, which allows us to grea...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Only peek at the tail element instead of popping it off, which allows
us to greatly simplify things when the tail element changes.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_recv_rtp_sink):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_sink_event):
Forward FLUSH events instead of leaking them.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the tail-changed callback in favour of a simple boolean when we
insert a buffer in the queue.
Add method to peek the tail of the buffer.
2007-10-05 12:07:37 +00:00
Wim Taymans
7067d01d2a gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. 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.
2007-09-20 14:34:57 +00:00
Wim Taymans
c0aa28ca5b gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(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_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2007-09-17 02:01:41 +00:00
Wim Taymans
6494828ef2 gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.
2007-09-15 18:48:03 +00:00
Wim Taymans
f6b7f47cf4 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.
2007-09-12 21:23:47 +00:00
Wim Taymans
79800df8b6 gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
(gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
Calculate and configure the NTP base time so that we can generate better
NTP times in SR packets.
Set caps on new ghostpad.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Clean debug statement.
* 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), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Add ntp-ns-base property to convert running_time to NTP time.
Handle NEWSEGMENT events on send and recv RTP pads so that we can
calculate the running time and thus NTP time of the packets.
Simplify getting the current NTP time using the pipeline clock.
Implement internal links functions.
Use the buffer timestamp to calculate the NTP time instead of the clock.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links),
(gst_rtp_ssrc_demux_src_query):
* gst/rtpmanager/gstrtpssrcdemux.h:
Implement internal links function.
Calculate the diff between different streams, this might be used later
to get the inter stream latency.
* gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
Simple cleanup.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
2007-09-12 18:04:32 +00:00
Tim-Philipp Müller
18b96b5233 gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
Make compiler happy: fix compilation with -Wall -Werror
(#473562).
2007-09-04 15:23:34 +00:00
Wim Taymans
fcce4aff92 gst/rtpmanager/: Updated example pipelines in docs.
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.
2007-09-03 21:19:34 +00:00
Wim Taymans
696bf74212 gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
2007-08-29 16:56:27 +00:00
Wim Taymans
9f597336b5 gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Distribute synchronisation parameters to the session manager so that it
can generate correct SR packets for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
(rtp_session_set_timestamp_sync), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Add methods for setting sync parameters.
Set correct RTP time in SR packets using the sync params.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Record last RTP <-> GST timestamp so that we can use them to convert NTP
to RTP timestamps in SR packets.
2007-08-29 01:22:43 +00:00
Wim Taymans
c0a64d008a gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
2007-08-28 20:30:16 +00:00
Tim-Philipp Müller
3adc3a9878 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.
2007-08-23 21:39:58 +00:00
Wim Taymans
6c781b9ca3 gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
2007-08-16 11:40:16 +00:00
Wim Taymans
9e50d836d4 gst/rtpmanager/: Remove complicated async queue and replace with more simple 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.
2007-08-10 17:16:53 +00:00
Wim Taymans
3a496fd7eb 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.
2007-05-28 16:37:47 +00:00
Wim Taymans
93888e03ac 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.
2007-05-23 13:08:52 +00:00
Wim Taymans
5171199836 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.
2007-04-30 13:41:30 +00:00
Wim Taymans
f4508d302c gst/rtpmanager/gstrtpsession.c: Remove debug.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
Remove debug.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_sdes), (calculate_rtcp_interval),
(rtp_session_next_timeout), (session_report_blocks):
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
Improve debugging
Fix interval for BYE/RTCP packets.
2007-04-29 14:46:27 +00:00
Wim Taymans
a468f02d2a gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
Move reconsideration code to the rtpsession object.
Simplify timout handling and add reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
(obtain_source), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_bye), (rtp_session_process_rtcp),
(calculate_rtcp_interval), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_start_rtcp),
(session_report_blocks), (session_cleanup), (session_sdes),
(session_bye), (is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Handle timeout of inactive sources and senders.
Implement BYE scheduling.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_process_sr), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add members to check for timeouts.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
(rtp_stats_calculate_bye_interval):
* gst/rtpmanager/rtpstats.h:
Use RFC algorithm for calculating the reporting interval.
2007-04-27 15:09:12 +00:00
Wim Taymans
f778bfb7cb gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Implement forward and reverse reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
(rtp_session_get_num_active_sources), (rtp_session_process_sr),
(session_report_blocks):
* gst/rtpmanager/rtpsession.h:
Small cleanups.
2007-04-25 16:38:03 +00:00
Wim Taymans
67c69ca0ea gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
2007-04-25 13:19:36 +00:00
Wim Taymans
34534179a2 gst/rtpmanager/gstrtpbin.c: fix for pad name change
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
2007-04-25 08:30:48 +00:00
Wim Taymans
1d75a69ccf configure.ac: Disable rtpmanager for now because it depends on CVS -base.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (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_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
2007-04-18 18:58:53 +00:00
Wim Taymans
92ef802b85 gst/rtpmanager/: Protect lists and structures with locks.
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.
2007-04-13 09:20:55 +00:00