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2009-03-13rtpbin: don't return FALSE on seek eventsWim Taymans1-0/+1
Silently ignore the seek event instead of returning FALSE.
2009-02-26gstrtpbin: Don't forward revc events to senderOlivier Crête1-4/+0
Don't send events from the receiver to the sender side. Fixes #572900.
2009-01-02gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use ↵Olivier Crete1-0/+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.
2009-01-02gst/rtpmanager/: Rename a method to better reflect what it really does.Wim Taymans1-1/+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.
2008-12-29gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.Wim Taymans1-2/+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.
2008-11-25gst/rtpmanager/gstrtpsession.c: Fix event leak.Wim Taymans1-1/+5
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src): Fix event leak.
2008-11-20gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when ↵Wim Taymans1-21/+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.
2008-10-16gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, ↵Wim Taymans1-0/+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.
2008-10-07gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.Håvard Graff1-6/+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.
2008-10-07gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.Wim Taymans1-0/+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.
2008-09-05gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender ↵Wim Taymans1-0/+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.
2008-08-28gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs ↵Wim Taymans1-3/+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.
2008-08-13gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we ↵Wim Taymans1-0/+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.
2008-07-03gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam ↵Peter Kjellerstedt1-7/+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.
2008-07-03gst/rtpmanager/: Do not mix the use of g_get_current_time() with ↵Peter Kjellerstedt1-28/+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().
2008-06-16gst/: More doc updates. More xrefs.Stefan Kost1-49/+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.
2008-06-12Do not use short_description in section docs for elements. We extract them ↵Stefan Kost1-1/+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.
2008-05-12gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned ↵Peter Kjellerstedt1-0/+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.
2008-05-08gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.Sjoerd Simons1-0/+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.
2008-04-25gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the ↵Wim Taymans1-0/+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.
2008-04-21gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.Olivier Crete1-6/+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.
2008-04-17gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a ↵Wim Taymans1-1/+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.
2008-04-09gst/rtpmanager/: Avoid leaking pads in the RTP manager.Peter Kjellerstedt1-0/+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.
2008-03-11gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP ↵Ole André Vadla Ravnås1-4/+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.
2008-01-11gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to ↵Wim Taymans1-4/+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.
2008-01-11gst/rtpmanager/: Make it possible to use different user_data for each of the ↵Youness Alaoui1-1/+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.
2008-01-09gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do ↵Wim Taymans1-1/+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.
2007-12-10gst/rtpmanager/: Post a message when the SDES infor changes for a source.Wim Taymans1-0/+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.
2007-12-10gst/rtpmanager/: Add signal to notify of an SDES change.Wim Taymans1-2/+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.
2007-12-10gst/rtpmanager/: Update comment.Wim Taymans1-6/+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.
2007-11-02gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).Ole André Vadla Ravnås1-3/+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).
2007-10-08gst/: Fix compiler warnings shown by Forte.Jan Schmidt1-3/+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.
2007-10-08gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.Wim Taymans1-0/+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.
2007-10-05gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead ↵Wim Taymans1-0/+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.
2007-09-20gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. ↵Wim Taymans1-0/+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.
2007-09-17gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one ↵Wim Taymans1-2/+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.
2007-09-15gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.Wim Taymans1-1/+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.
2007-09-12gst/rtpmanager/: Various leak fixes.Wim Taymans1-0/+2
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.
2007-09-12gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so ↵Wim Taymans1-15/+123
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.
2007-09-04gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with ↵Tim-Philipp Müller1-2/+3
-Wall -Werror (#473562). Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
2007-09-03gst/rtpmanager/: Updated example pipelines in docs.Wim Taymans1-78/+151
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.
2007-08-29gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.Wim Taymans1-7/+63
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.
2007-08-29gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the ↵Wim Taymans1-1/+45
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.
2007-08-28gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.Wim Taymans1-3/+6
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.
2007-08-23Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers ↵Tim-Philipp Müller1-66/+66
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.
2007-08-16gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.Wim Taymans1-2/+16
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.
2007-08-10gst/rtpmanager/: Remove complicated async queue and replace with more simple ↵Wim Taymans1-0/+132
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.
2007-05-28Rename elements to avoid conflict with farsight elements with the same name. ↵Wim Taymans1-17/+17
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.
2007-05-23Document stuff.Wim Taymans1-5/+117
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.
2007-04-30gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns ↵Wim Taymans1-6/+10
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.