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2008-04-21gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.Olivier Crete4-14/+41
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 Taymans4-5/+28
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 Kjellerstedt2-0/+24
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/rtpsession.*: Implement collision and loop detection in ↵Olivier Crete4-22/+253
rtpmanager. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses), (check_collision), (obtain_source), (rtp_session_create_new_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_send_bye_locked), (rtp_session_send_bye), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Implement collision and loop detection in rtpmanager. Fixes #520626. * gst/rtpmanager/rtpsource.c: (rtp_source_reset), (rtp_source_init): * gst/rtpmanager/rtpsource.h: Add method to reset stats.
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.
2008-03-11gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we ↵Stefan Kost1-1/+1
have no timestamps. Original commit message from CVS: Patch by: Stefan Kost <ensonic@users.sf.net> * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Don't try to reset the clock skew when we have no timestamps. Fixes #519005.
2008-02-20gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug ↵Olivier Crete1-0/+2
517571. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Fix small memory leak, leaking caps. Fixes #bug 517571.
2008-02-14gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet ↵Olivier Crete1-1/+4
when doing synchronisation. Fixes #516160. Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate): Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
2008-01-29gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the ↵Thijs Vermeir1-0/+7
buffer caps when we receive a new payload... Original commit message from CVS: Patch by: Thijs Vermeir <thijsvermeir at gmail dot com> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Try to get the new clock-rate from the buffer caps when we receive a new payload type instead of always firing the signal. Fixes #512774.
2008-01-25gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not ↵Olivier Crete1-3/+47
provided with caps but with a signal. Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (create_stream), (payload_type_change), (new_ssrc_pad_found): Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
2008-01-25gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and ↵Olivier Crete3-32/+11
extend it so that a clock-rate can be provided... Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided with each buffer instead. Fixes #511686.
2008-01-25gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.Olivier Crete1-6/+9
Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove old unused variable. Track pt on input buffers and get the clock-rate when it changes. Ignore packets with unknown clock-rate. See #511686.
2008-01-25gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. ↵Olivier Crete1-1/+1
Fixes #511920 Original commit message from CVS: Patch by: Olivier Crete <tester@tester.ca> * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920
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 Alaoui3-22/+174
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.
2008-01-10gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patchThijs Vermeir1-2/+2
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
2008-01-10gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)Thijs Vermeir1-4/+46
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
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.
2008-01-04gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we ↵Wim Taymans1-3/+1
don't have ownership. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Don't unref the popped buffer when we don't have ownership. Fixes #507020.
2007-12-31gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.Wim Taymans1-0/+1
Original commit message from CVS: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_change_state): Don't clean up pads when going to PAUSED.
2007-12-12gst/rtpmanager/: Clean up the dynamic pads when going to READY.Wim Taymans2-30/+41
Original commit message from CVS: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset), (gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query), (gst_rtp_ssrc_demux_change_state): Clean up the dynamic pads when going to READY.
2007-12-12gst/rtpmanager/: Fix some leaks.Wim Taymans3-0/+17
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_handle_message): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize), (rtp_session_send_bye): * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Fix some leaks.
2007-12-10gst/rtpmanager/: Post a message when the SDES infor changes for a source.Wim Taymans4-3/+118
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 Taymans20-38/+85
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/gstrtpbin.*: Expose SDES items as properties and configure ↵Wim Taymans3-2/+200
the session managers with them. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose SDES items as properties and configure the session managers with them. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_set_property): Fix SSRC property.
2007-12-10gst/rtpmanager/: Update comment.Wim Taymans7-322/+996
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.
2007-11-22gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited ↵Wim Taymans1-4/+2
amount of time and thus has no max_latency ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): jitterbuffer can buffer an unlimited amount of time and thus has no max_latency requirements.
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).
2007-10-09gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.Laurent Glayal1-0/+1
Original commit message from CVS: Patch by: Laurent Glayal <spglegle at yahoo dot fr> * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_class_init): Fix memleak. Fixes #484990.
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 Taymans4-46/+90
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/rtpsession.c: When reconsidering RTCP timeouts, set the next ↵Wim Taymans1-11/+32
timeout against the last report time inst... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout), When reconsidering RTCP timeouts, set the next timeout against the last report time instead of the current clock time so that we don't end up reconsidering forever.
2007-10-05gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead ↵Wim Taymans5-46/+35
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.
2007-10-02gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables.Wim Taymans3-29/+73
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (apply_offset), (gst_rtp_jitter_buffer_loop): Remove some old unused variables. Don't add the latency to the skew corrected timestamp, latency is only used to sync against the clock. Improve debugging. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_reset_skew), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Handle case where server timestamp goes backwards or wildly jumps by temporarily pausing the skew correction. Improve debugging.
2007-09-28gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.Wim Taymans2-7/+8
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (free_client): Fix crasher in dispose. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Handle cases where input buffers have no timestamps so that no clock skew can be calculated, in this case interpollate timestamps based on rtp timestamp and assume a 0 clock skew.
2007-09-28gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now ↵Wim Taymans3-92/+105
in the lower level object. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): Remove jitter correction code, it's now in the lower level object. Use new -core method for doing a peer query. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Move jitter correction to the lowlevel jitterbuffer. Increase the max window size. When filling the window, already start estimating the skew using a parabolic weighting factor so that we have a much better startup behaviour that gets more accurate with the more samples we have. Increase the default weighting factor for the steady state to get smoother timestamps.
2007-09-26gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.Wim Taymans3-7/+15
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): Fix cleanup crasher. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Dynamically adjust the skew calculation window so that we calculate it over a period of around 2 seconds.
2007-09-20gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. ↵Wim Taymans6-0/+74
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 Taymans6-24/+56
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.
2007-09-16gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.Wim Taymans6-139/+341
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): Use lock to protect variable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop): Reconstruct GST timestamp from RTP timestamps based on measured clock skew and sync offset. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_set_tail_changed), (rtp_jitter_buffer_set_clock_rate), (rtp_jitter_buffer_get_clock_rate), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek): * gst/rtpmanager/rtpjitterbuffer.h: Measure clock skew. Add callback to be notfied when a new packet was inserted at the tail. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Remove clock skew detection, it's move to the jitterbuffer now.
2007-09-15gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.Wim Taymans3-6/+44
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 Taymans7-12/+108
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 Taymans8-39/+306
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 Taymans12-435/+1306
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.
2007-08-31gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release ↵Wim Taymans1-13/+23
buffers from the jitterbuffer so that we can h... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop): Use extended timestamp to release buffers from the jitterbuffer so that we can handle the rtp wraparound correctly.
2007-08-29gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.Wim Taymans3-10/+69
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 Taymans6-8/+145
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 Taymans4-7/+68
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.
2007-08-27gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer ↵Wim Taymans1-1/+14
latency into account. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): When synchronizing buffers, take peer latency into account. Don't try to add our latency to invalid peer max latency values.