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2008-11-19gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to ↵Wim Taymans1-4/+3
get lip-sync notifications. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_handle_sync), (create_stream), (free_stream), (new_ssrc_pad_found): Remove internal sync pad, use signals instead to get lip-sync notifications. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink), (remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad), (gst_rtp_jitter_buffer_release_pad), (gst_rtp_jitter_buffer_sink_rtcp_event), (gst_rtp_jitter_buffer_chain_rtcp), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: Make it possible to send SR packets to the jitterbuffer. Check if the SR timestamps are valid by comparing them to the RTP timestamps. Signal the SR packet and the timing information to listeners. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query): Remove some unused code. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of the last seen RTP timestamp so that we can filter out invalid SR packets.
2008-11-17gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ↵Wim Taymans1-1/+2
ourselves but simply get the value from the ji... Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found): Do not try to keep track of the clock-rate ourselves but simply get the value from the jitterbuffer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add some debug info. Pass the clock-rate to the jitterbuffer. Also pass the clock-rate along with the rtp timestamp when getting the sync parameters. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Fix some debug. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of clock-rate changes and return the clock-rate together with the rtp timestamps used for sync. Don't try to construct timestamps when we have no base_time. * gst/rtpmanager/rtpsource.c: (get_clock_rate): Request a new clock-rate when the payload type changes. Reset the jitter calculation when the clock-rate changes.
2008-09-05gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender ↵Wim Taymans1-0/+3
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.
2007-12-10gst/rtpmanager/: Add signal to notify of an SDES change.Wim Taymans1-1/+1
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.
2007-08-23Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers ↵Tim-Philipp Müller1-11/+11
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.
2007-05-23Document stuff.Wim Taymans1-1/+8
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.
2007-04-18configure.ac: Disable rtpmanager for now because it depends on CVS -base.Wim Taymans1-0/+1
Original commit message from CVS: * configure.ac: Disable rtpmanager for now because it depends on CVS -base. * gst/rtpmanager/Makefile.am: Added new files for session manager. * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (pt_map_requested), (new_ssrc_pad_found): Some cleanups. the session manager can now also request a pt-map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate), (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpsession.h: We can ask for pt-map now too when the session manager needs it. Hook up to the new session manager, implement the needed callbacks for pushing data, getting clock time and requesting clock-rates. Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to be send to clients. Add code to start and stop the thread that will schedule RTCP through the session manager. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth), (source_push_rtp), (source_clock_rate), (check_collision), (obtain_source), (rtp_session_add_source), (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_get_source_by_ssrc), (rtp_session_get_source_by_cname), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_process_app), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_rtcp_interval), (rtp_session_produce_rtcp): * gst/rtpmanager/rtpsession.h: The advanced beginnings of the main session manager that handles the participant database of RTPSources, SSRC probation, SSRC collisions, parse RTCP to update source stats. etc.. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_finalize), (rtp_source_new), (rtp_source_set_callbacks), (rtp_source_set_as_csrc), (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from), (push_packet), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_process_bye), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb): * gst/rtpmanager/rtpsource.h: Object that encapsulates an SSRC and its state in the database. Calculates the jitter and transit times of data packets. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter): * gst/rtpmanager/rtpstats.h: Various stats regarding the session and sources. Used to calculate the RTCP interval.
2007-04-12gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.Wim Taymans1-1/+1
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested): Emit pt map requests and cache results. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_get_clock_rate), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Emit request-pt-map signals.
2007-04-11gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.Wim Taymans1-0/+2
Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers. * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (clock_rate_request), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp): * gst/rtpmanager/gstrtpbin.h: Prepare for caching pt maps. Connect to signals to collect pt maps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: Add request_clock_rate signal. Use scale insteat of scale_int because the later does not deal with negative numbers. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_chain): * gst/rtpmanager/gstrtpptdemux.h: Implement request-pt-map signal.
2007-04-03Add RTP session management elements. Still in progress.Wim Taymans1-0/+74
Original commit message from CVS: * configure.ac: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new), (signal_waiting_threads), (async_jitter_queue_ref), (async_jitter_queue_ref_unlocked), (async_jitter_queue_set_low_threshold), (async_jitter_queue_set_high_threshold), (async_jitter_queue_set_max_queue_length), (async_jitter_queue_get_g_queue), (calculate_ts_diff), (async_jitter_queue_length_ts_units_unlocked), (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref), (async_jitter_queue_lock), (async_jitter_queue_unlock), (async_jitter_queue_push), (async_jitter_queue_push_unlocked), (async_jitter_queue_push_sorted), (async_jitter_queue_push_sorted_unlocked), (async_jitter_queue_insert_after_unlocked), (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop), (async_jitter_queue_pop_unlocked), (async_jitter_queue_length), (async_jitter_queue_length_unlocked), (async_jitter_queue_set_flushing_unlocked), (async_jitter_queue_unset_flushing_unlocked), (async_jitter_queue_set_blocking_unlocked): * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property), (gst_rtp_bin_change_state), (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream), (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init), (gst_rtp_client_class_init), (gst_rtp_client_init), (gst_rtp_client_finalize), (gst_rtp_client_set_property), (gst_rtp_client_get_property), (gst_rtp_client_change_state), (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad): * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps), (gst_jitter_buffer_sink_setcaps), (free_func), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_activate_push), (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt), (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init), (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init), (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain), (gst_rtp_pt_demux_getcaps), (find_pad_for_pt), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (gst_rtp_session_change_state), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad): * gst/rtpmanager/gstrtpsession.h: Add RTP session management elements. Still in progress.