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* rtpmanager: Fixed a copy & paste errorPeter Kjellerstedt2009-09-011-2/+2
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* rtpmanager: Removed unused variable privPeter Kjellerstedt2009-09-011-22/+0
| | | | | The variable priv was initialized in a lot of functions but then never used for anything.
* rtpmanager: A little clean upPeter Kjellerstedt2009-09-011-3/+4
| | | | | Make the code flow of gst_rtp_session_send_rtcp() and gst_rtp_session_sync_rtcp() identical.
* rtpmanager: Make sure that used caps are not freed already (take 2)Peter Kjellerstedt2009-09-011-2/+4
| | | | | | This reintroduces the fix for bug #593391. It also applies it in gst_rtp_session_sync_rtcp() which has very similar code to gst_rtp_session_send_rtcp().
* rtpmanager: Set caps in gst_rtp_session_send_rtcp() correctly againPeter Kjellerstedt2009-09-011-1/+1
| | | | | The test for when to set an RTCP caps on the output pad in gst_rtp_session_send_rtcp() accidentally got inverted in the last commit.
* rtpsession: use proper locking for pads and capsWim Taymans2009-08-311-34/+85
| | | | | | | Use the sesion lock and shotdown variable to protect and ref the pads we are going to push on. fixes #561825
* rtpsession: Make sure that used caps are not freed alreadySebastian Dröge2009-08-311-2/+5
| | | | Fixes bug #593391.
* rtp: Use new gst_iterator_new_single() for the internal linked pads iterationSebastian Dröge2009-08-311-71/+15
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* rtpsession: Use iterate internal links instead of deprecated get internal linksSebastian Dröge2009-08-311-29/+87
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* rtpbin: add SDES propertyWim Taymans2009-08-111-102/+10
| | | | | | 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.
* rtpbin: add support for buffer-listBranko Subasic2009-08-111-14/+48
| | | | | | | Add support for sending buffer-lists. Add unit test for testing that the buffer-list passed through rtpbin. fixes #585839
* rtpsession: reuse source code for SDESWim Taymans2009-08-111-58/+2
| | | | Reuse the RTPSource object property instead of duplicating code.
* rtpsession: join the RTCP threadWim Taymans2009-08-111-0/+5
| | | | | | Avoid a case where a joinable thread would be left unjoined, which leaked the thread structure. Fixes #577318.
* rtpbin: don't return FALSE on seek eventsWim Taymans2009-08-111-0/+1
| | | | Silently ignore the seek event instead of returning FALSE.
* gstrtpbin: Don't forward revc events to senderOlivier Crête2009-08-111-4/+0
| | | | | Don't send events from the receiver to the sender side. Fixes #572900.
* gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use ↵Olivier Crete2009-08-111-0/+23
| | | | | | | | | | | | this as the internal SSRC. Fixes #565910. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink): * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc): When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
* gst/rtpmanager/: Rename a method to better reflect what it really does.Wim Taymans2009-08-111-1/+1
| | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_getcaps_send_rtp): * gst/rtpmanager/rtpsession.c: (check_collision), (rtp_session_schedule_bye_locked), (rtp_session_schedule_bye): * gst/rtpmanager/rtpsession.h: Rename a method to better reflect what it really does.
* gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.Wim Taymans2009-08-111-2/+4
| | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp): Use method to get the internal SSRC. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_set_property), (rtp_session_get_property): Add property to congiure the internal SSRC of the session. Fixes #565910.
* gst/rtpmanager/gstrtpsession.c: Fix event leak.Wim Taymans2009-08-111-1/+5
| | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src): Fix event leak.
* gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when ↵Wim Taymans2009-08-111-21/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | processing RTP packets. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (get_current_times), (rtcp_thread), (gst_rtp_session_chain_recv_rtp): Pass the running time to the session when processing RTP packets. Improve the time function to provide more info. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sdes), (rtp_session_process_rtcp), (session_start_rtcp), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Mark the internal source with a flag. Use running_time instead of the more useless timestamp. Validate a source when a valid SDES has been received. Pass the current system time when processing SR packets. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_create_stats), (rtp_source_get_property), (rtp_source_send_rtp), (rtp_source_process_rb), (rtp_source_get_new_rb), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add property to get source stats. Mark params as STATIC_STRINGS. Calculate the bitrate at the sender SSRC. Avoid negative values in the round trip time calculations. * gst/rtpmanager/rtpstats.h: Update some docs and change some variable name to more closely reflect what it contains.
* gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, ↵Wim Taymans2009-08-111-0/+29
| | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.Håvard Graff2009-08-111-6/+9
| | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.Wim Taymans2009-08-111-0/+102
| | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender ↵Wim Taymans2009-08-111-0/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs ↵Wim Taymans2009-08-111-3/+13
| | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we ↵Wim Taymans2009-08-111-0/+7
| | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam ↵Peter Kjellerstedt2009-08-111-7/+7
| | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Do not mix the use of g_get_current_time() with ↵Peter Kjellerstedt2009-08-111-28/+25
| | | | | | | | | | | | | | | | | | | 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().
* gst/: More doc updates. More xrefs.Stefan Kost2009-08-111-49/+30
| | | | | | | | | | | | | 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.
* Do not use short_description in section docs for elements. We extract them ↵Stefan Kost2009-08-111-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned ↵Peter Kjellerstedt2009-08-111-0/+3
| | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.Sjoerd Simons2009-08-111-0/+1
| | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the ↵Wim Taymans2009-08-111-0/+33
| | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.Olivier Crete2009-08-111-6/+19
| | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a ↵Wim Taymans2009-08-111-1/+4
| | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Avoid leaking pads in the RTP manager.Peter Kjellerstedt2009-08-111-0/+9
| | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP ↵Ole André Vadla Ravnås2009-08-111-4/+36
| | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to ↵Wim Taymans2009-08-111-4/+3
| | | | | | | | | 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.
* gst/rtpmanager/: Make it possible to use different user_data for each of the ↵Youness Alaoui2009-08-111-1/+4
| | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do ↵Wim Taymans2009-08-111-1/+0
| | | | | | | | | 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.
* gst/rtpmanager/: Post a message when the SDES infor changes for a source.Wim Taymans2009-08-111-0/+66
| | | | | | | | | | | | 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.
* gst/rtpmanager/: Add signal to notify of an SDES change.Wim Taymans2009-08-111-2/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Update comment.Wim Taymans2009-08-111-6/+162
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).Ole André Vadla Ravnås2009-08-111-3/+3
| | | | | | | 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).
* gst/: Fix compiler warnings shown by Forte.Jan Schmidt2009-08-111-3/+3
| | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.Wim Taymans2009-08-111-0/+1
| | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead ↵Wim Taymans2009-08-111-0/+1
| | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. ↵Wim Taymans2009-08-111-0/+25
| | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one ↵Wim Taymans2009-08-111-2/+25
| | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.Wim Taymans2009-08-111-1/+33
| | | | | | | | | | | | | | | | | 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.
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