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    • Wim Taymans's avatar
      rtpbin: remove more ntpnstime and cleanups · 5a4ecc9d
      Wim Taymans authored
      Remove some code where we pass ntpnstime around, we can do most things with the
      running_time just fine.
      Rename a variable in the ArrivalStats struct so that it's clear that this is the
      current system time.
      5a4ecc9d
  21. 11 Aug, 2009 9 commits
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets. · 2f5b130a
      Wim Taymans authored
      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.
      2f5b130a
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver. · 85e26f65
      Wim Taymans authored
      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.
      85e26f65
    • Wim Taymans's avatar
      gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first · e779adca
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
      (gst_rtp_bin_sync_chain):
      * gst/rtpmanager/rtpsession.c: (update_arrival_stats),
      (rtp_session_process_sr), (rtp_session_on_timeout):
      * gst/rtpmanager/rtpsource.c: (rtp_source_init),
      (calculate_jitter):
      * gst/rtpmanager/rtpsource.h:
      * gst/rtpmanager/rtpstats.h:
      Also keep track of the first buffer timestamp together with the first
      RTP timestamp as they both are needed to construct the timing of
      outgoing packets in the jitterbuffer and are therefore also needed to
      manage lip-sync. This fixes lip-sync if the first RTP packets arrive
      with a wildly different gap.
      e779adca
    • Wim Taymans's avatar
      gst/rtpmanager/: Add signal to notify of an SDES change. · 49e501a6
      Wim Taymans authored
      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.
      49e501a6
    • Wim Taymans's avatar
      gst/rtpmanager/: Updated example pipelines in docs. · e7b6212c
      Wim Taymans authored
      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.
      e7b6212c
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object. · a7b80281
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
      (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
      Move reconsideration code to the rtpsession object.
      Simplify timout handling and add reconsideration.
      * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
      (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
      (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
      (obtain_source), (rtp_session_create_source),
      (update_arrival_stats), (rtp_session_process_rtp),
      (rtp_session_process_sr), (rtp_session_process_rr),
      (rtp_session_process_bye), (rtp_session_process_rtcp),
      (calculate_rtcp_interval), (rtp_session_send_bye),
      (rtp_session_next_timeout), (session_start_rtcp),
      (session_report_blocks), (session_cleanup), (session_sdes),
      (session_bye), (is_rtcp_time), (rtp_session_on_timeout):
      * gst/rtpmanager/rtpsession.h:
      Handle timeout of inactive sources and senders.
      Implement BYE scheduling.
      * gst/rtpmanager/rtpsource.c: (calculate_jitter),
      (rtp_source_process_sr), (rtp_source_get_last_sr),
      (rtp_source_get_last_rb):
      * gst/rtpmanager/rtpsource.h:
      Add members to check for timeouts.
      * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
      (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
      (rtp_stats_calculate_bye_interval):
      * gst/rtpmanager/rtpstats.h:
      Use RFC algorithm for calculating the reporting interval.
      a7b80281
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED. · ae536e0c
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpjitterbuffer.c:
      (gst_rtp_jitter_buffer_change_state):
      Report NO_PREROLL when going to PAUSED.
      * gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
      Don't send RTCP right before we are shutting down.
      * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
      (rtp_session_process_sr), (session_report_blocks),
      (rtp_session_perform_reporting):
      Improve report blocks.
      * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
      (rtp_source_process_rtp), (rtp_source_process_sr),
      (rtp_source_process_rb), (rtp_source_get_last_sr),
      (rtp_source_get_last_rb):
      * gst/rtpmanager/rtpsource.h:
      * gst/rtpmanager/rtpstats.h:
      Cleanups, add methods to access stats.
      ae536e0c
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpbin.c: fix for pad name change · 23883be0
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpbin.c: (create_rtcp):
      fix for pad name change
      * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
      (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
      Fix for renamed methods.
      * gst/rtpmanager/rtpsession.c: (rtp_session_init),
      (rtp_session_finalize), (rtp_session_set_cname),
      (rtp_session_get_cname), (rtp_session_set_name),
      (rtp_session_get_name), (rtp_session_set_email),
      (rtp_session_get_email), (rtp_session_set_phone),
      (rtp_session_get_phone), (rtp_session_set_location),
      (rtp_session_get_location), (rtp_session_set_tool),
      (rtp_session_get_tool), (rtp_session_set_note),
      (rtp_session_get_note), (source_push_rtp), (obtain_source),
      (rtp_session_add_source), (rtp_session_get_source_by_ssrc),
      (rtp_session_create_source), (rtp_session_process_rtp),
      (rtp_session_process_sr), (rtp_session_process_sdes),
      (rtp_session_process_rtcp), (rtp_session_send_rtp),
      (rtp_session_get_reporting_interval), (session_report_blocks),
      (session_sdes), (rtp_session_perform_reporting):
      * gst/rtpmanager/rtpsession.h:
      Prepare for implementing SSRC sampling.
      Create SSRC for the session.
      Add methods to set the SDES entries.
      fix accounting of senders/receivers.
      Implement SR/RR/SDES RTCP reporting.
      * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
      (rtp_source_process_rtp), (rtp_source_process_sr):
      * gst/rtpmanager/rtpsource.h:
      Implement extended sequence number.
      * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
      * gst/rtpmanager/rtpstats.h:
      Rename some fields.
      23883be0
    • Wim Taymans's avatar
      configure.ac: Disable rtpmanager for now because it depends on CVS -base. · 54b3dec1
      Wim Taymans authored
      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.
      54b3dec1