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* rtpbin: don't return FALSE on seek eventsWim Taymans2009-08-111-0/+1
| | | | Silently ignore the seek event instead of returning FALSE.
* gstrtpbin: Don't forward revc events to senderOlivier Crête2009-08-111-4/+0
| | | | | Don't send events from the receiver to the sender side. Fixes #572900.
* docs: various doc fixesStefan Kost2009-08-111-1/+1
| | | | | | No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please.
* Send BYE packets immediatly for small sessionsWim Taymans2009-08-111-0/+4
| | | | | | When the number of participants is less than 50, the RFC allows for sending the BYE packet immediatly instead of using the regular BYE timeout. Fixes #567828.
* Unlock the jitterbuffer before pushing out the packet-lost events.Wim Taymans2009-08-111-5/+8
| | | | | Move some code before we do the unlock to make the jitterbuffer state consistent while we are unlocked.
* gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use ↵Olivier Crete2009-08-112-0/+28
| | | | | | | | | | | | this as the internal SSRC. Fixes #565910. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink): * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc): When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
* gst/rtpmanager/: Rename a method to better reflect what it really does.Wim Taymans2009-08-113-19/+9
| | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_getcaps_send_rtp): * gst/rtpmanager/rtpsession.c: (check_collision), (rtp_session_schedule_bye_locked), (rtp_session_schedule_bye): * gst/rtpmanager/rtpsession.h: Rename a method to better reflect what it really does.
* gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.Wim Taymans2009-08-112-2/+16
| | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp): Use method to get the internal SSRC. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_set_property), (rtp_session_get_property): Add property to congiure the internal SSRC of the session. Fixes #565910.
* gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset ↵Wim Taymans2009-08-111-6/+8
| | | | | | | | | the internal source when the SSRC actually... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc): Only change the SSRC of the session and reset the internal source when the SSRC actually changed. See #565910.
* gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but ↵Wim Taymans2009-08-112-4/+18
| | | | | | | | | | | | there was a clock-rate, assume the clock-ra... Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate): * gst/rtpmanager/rtpsource.h: When no payload was specified on the caps but there was a clock-rate, assume the clock-rate corresponds to the first payload type found in the RTP packets. Fixes #565509.
* gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp ↵Arnout Vandecappelle2009-08-112-2/+32
| | | | | | | | | | | | | | and of the last sender-side time. Timest... Original commit message from CVS: Patch by: Arnout Vandecappelle <arnout at mind dot be> * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of the last outgoing timestamp and of the last sender-side time. Timestamps can only go forward if they do at the sender side, can only go back if they do at the sender side, and remain the same if they remain the same at the sender side. Fixes #565319.
* gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so ↵Wim Taymans2009-08-111-9/+18
| | | | | | | | | | | | | | that we don't lose our ref to it when a se... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (obtain_source), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye): Make obtain_source return an aditional ref so that we don't lose our ref to it when a session cleanup occurs when we are emiting a signal. Emit the on_new_ssrc signal for the CSRC, not the SSRC. Fixes #562319.
* gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the ↵Wim Taymans2009-08-111-24/+27
| | | | | | | | | | payload type map too. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync), (gst_rtp_bin_clear_pt_map): Reset the sync parameters when clearing the payload type map too. Fixes #562312.
* gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not ↵Wim Taymans2009-08-112-66/+67
| | | | | | | | | | | | | | | | needed and pass new info in the method call. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_client), (gst_rtp_bin_reset_sync), (gst_rtp_bin_associate), (gst_rtp_bin_handle_sync), (create_stream), (gst_rtp_bin_class_init), (new_ssrc_pad_found): * gst/rtpmanager/gstrtpbin.h: Remove a lot of per stream state that is not needed and pass new info in the method call. Add signal to reset sync parameters. Avoid parsing the caps to get a clock_base, we get this from the sync signal now.
* gst/rtpmanager/gstrtpsession.c: Fix event leak.Wim Taymans2009-08-111-1/+5
| | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src): Fix event leak.
* gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU.Wim Taymans2009-08-111-1/+15
| | | | | | | | Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_set_property), (rtp_session_get_property): Add property to configure the RTCP MTU.
* gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS.Wim Taymans2009-08-112-120/+141
| | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (copy_source), (rtp_session_create_sources), (rtp_session_get_property): Add G_PARAM_STATIC_STRINGS. Add property to return a GValueArray of all known RTPSources in the session. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_create_sdes), (rtp_source_set_property), (rtp_source_get_property): Remove properties to set the various SDES items, an application is never supposed to change the RTPSource data. Change the SDES getter properties to one SDES property that returns all SDES items in a GstStructure.
* gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads.Wim Taymans2009-08-111-1/+1
| | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad): Also unref the target pad for unknown pads.
* gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752.Olivier Crete2009-08-111-6/+12
| | | | | | | Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad): Release the right pads on rtpbin. Fixes #561752.
* gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when ↵Wim Taymans2009-08-116-60/+213
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | processing RTP packets. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (get_current_times), (rtcp_thread), (gst_rtp_session_chain_recv_rtp): Pass the running time to the session when processing RTP packets. Improve the time function to provide more info. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sdes), (rtp_session_process_rtcp), (session_start_rtcp), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Mark the internal source with a flag. Use running_time instead of the more useless timestamp. Validate a source when a valid SDES has been received. Pass the current system time when processing SR packets. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_create_stats), (rtp_source_get_property), (rtp_source_send_rtp), (rtp_source_process_rb), (rtp_source_get_new_rb), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add property to get source stats. Mark params as STATIC_STRINGS. Calculate the bitrate at the sender SSRC. Avoid negative values in the round trip time calculations. * gst/rtpmanager/rtpstats.h: Update some docs and change some variable name to more closely reflect what it contains.
* gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler ↵Sebastian Dröge2009-08-111-1/+1
| | | | | | | | | | warning about uninitialized variable. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain_rtcp): Initialize return value to fix compiler warning about uninitialized variable.
* gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope.Wim Taymans2009-08-111-1/+1
| | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init): Mark signal arg as static scope.
* gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to ↵Wim Taymans2009-08-116-139/+382
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many ↵Sebastian Dröge2009-08-111-1/+1
| | | | | | | | | arguments as required by the format string. Fixes... Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (get_clock_rate): Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes a compiler warning.
* gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ↵Wim Taymans2009-08-117-62/+48
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtpmanager/: Small cleanups and some more debug info.Wim Taymans2009-08-112-7/+13
| | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain): * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew): Small cleanups and some more debug info.
* gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output ↵Wim Taymans2009-08-111-5/+5
| | | | | | | | | | seqnum when we get a seqnum-base on the ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain): Also configure the next expected output seqnum when we get a seqnum-base on the caps.
* Don't install static libs for plugins. Fixes #550851 for -bad.Stefan Kost2009-08-111-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad.
* gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output ↵Wim Taymans2009-08-112-28/+46
| | | | | | | | | | | | | | | | | seqnum counter to check for input seqnum disco... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Fix problem with using the output seqnum counter to check for input seqnum discontinuities. Improve gap detection and recovery, reset and flush the jitterbuffer on seqnum restart. Fixes #556520. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert): Fix wrong G_LIKELY.
* gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, ↵Wim Taymans2009-08-111-0/+29
| | | | | | | | | | make LATENCY event return Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src): Install event handler on the rtcp_src pad, make LATENCY event return TRUE.
* gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.Håvard Graff2009-08-116-16/+85
| | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Håvard Graff <havard dot graff at tandberg dot com> * gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Add action signal to retrieve the internal RTPSession object. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_get_property), (gst_rtp_session_release_pad): Add property to access the internal RTPSession object. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (check_collision): * gst/rtpmanager/rtpsession.h: Add action signal to retrieve an RTPSource object by SSRC. See #555396.
* gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.Wim Taymans2009-08-112-4/+197
| | | | | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_pad), (free_session), (gst_rtp_bin_dispose), (remove_recv_rtp), (remove_recv_rtcp), (remove_send_rtp), (remove_rtcp), (gst_rtp_bin_release_pad): Release pads of the session manager. Start implementing releasing pads of gstrtpbin. * gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink), (remove_recv_rtcp_sink), (remove_send_rtp_sink), (remove_send_rtcp_src), (gst_rtp_session_release_pad): Implement releasing pads in gstrtpsession.
* gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was ↵Wim Taymans2009-08-111-5/+7
| | | | | | | | | | not already configured for the streams. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps): Only update the seqnum-base when it was not already configured for the streams.
* gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the ↵Wim Taymans2009-08-111-0/+18
| | | | | | | | | | | session lock when we emit the signals. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout): Ref the rtpsource object before we release the session lock when we emit the signals.
* gst/rtpmanager/: Fix some docs.Wim Taymans2009-08-113-1/+4
| | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpsession.c: (on_sender_timeout), (session_cleanup): * gst/rtpmanager/rtpsource.c: Fix some docs.
* Fix compiler warnings on OS/XJan Schmidt2009-08-111-1/+1
| | | | | | | Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_process_cb): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Fix compiler warnings on OS/X
* gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for ↵Wim Taymans2009-08-111-7/+34
| | | | | | | | | | which we have not yet seen an SR packet. A... Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain): Do not try to adjust the offset of streams for which we have not yet seen an SR packet. Avoids large ts-offsets in some cases.
* gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender ↵Wim Taymans2009-08-1113-107/+218
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | becomes a receiver. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout), (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: Add signal to notify listeners when a sender becomes a receiver. Tweak lip-sync code, don't store our own copy of the ts-offset of the jitterbuffer, don't adjust sync if the change is less than 4msec. Get the RTP timestamp <-> GStreamer timestamp relation directly from the jitterbuffer instead of our inaccurate version from the source. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add G_LIKELY macros, use global defines for max packet reorder and dropouts. Reset the jitterbuffer clock skew detection when packets seqnums are changed unexpectedly. * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: Add sender timeout signal. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Add some G_LIKELY macros. Keep track of the extended RTP timestamp so that we can report the RTP timestamp <-> GStreamer timestamp relation for lip-sync. Remove server timestamp gap detection code, the server can sometimes make a huge gap in timestamps (talk spurts,...) see #549774. Detect timetamp weirdness instead by observing the sender/receiver timestamp relation and resync if it changes more than 1 second. Add method to report about the current rtp <-> gst timestamp relation which is needed for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_sender_timeout), (check_collision), (rtp_session_process_sr), (session_cleanup): * gst/rtpmanager/rtpsession.h: Add sender timeout signal. Remove inaccurate rtp <-> gst timestamp relation code, the jitterbuffer can now do an accurate reporting about this. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (calculate_jitter), (rtp_source_process_rtp): * gst/rtpmanager/rtpsource.h: Remove inaccurate rtp <-> gst timestamp relation code. * gst/rtpmanager/rtpstats.h: Define global max-reorder and max-dropout constants for use in various subsystems.
* gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs ↵Wim Taymans2009-08-113-6/+22
| | | | | | | | | | | | | | | | us to. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp), (gst_rtp_session_event_send_rtp_sink): Send EOS when the session object instructs us to. * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Make it possible for the session manager to instruct us to send EOS. We currently will EOS when the session is a sender and when the sender part goes EOS. This is not entirely correct behaviour because the session could still participate as a receiver. Fixes #549409.
* gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we ↵Wim Taymans2009-08-116-8/+88
| | | | | | | | | | | | | | | | | | | | | | detect a gap when the clock_base changed. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (new_ssrc_pad_found): Reset rtp timestamp interpollation when we detect a gap when the clock_base changed. Don't try to adjust the ts-offset when it's too big (> 3seconds) * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc): * gst/rtpmanager/gstrtpsession.h: Add method to set session SSRC. * gst/rtpmanager/rtpsession.c: (check_collision), (rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Added debugging for the collision checks. Add method to change the internal SSRC of the session. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Reset the clock base when we detect large jumps in the seqnums.
* gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log.Stefan Kost2009-08-111-1/+2
| | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log. * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.c: Use "-" instead of "_" in property names. Can we call them just "device" like everywhere else?
* gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable ↵Olivier Crete2009-08-113-2/+13
| | | | | | | | | | | | | | | | | | before inserting it in the jitterbuffer becaus... Original commit message from CVS: Based on patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Make the buffer metadata writable before inserting it in the jitterbuffer because the jitterbuffer will modify the timestamps. * gst/rtpmanager/rtpjitterbuffer.c: Update method comment about requiring writable metadata on buffers. * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr), (rtp_session_process_rtcp): Make the RTCP buffer metadata writable because we want to modify the metadata. Fixes #546312.
* gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.Håvard Graff2009-08-111-2/+3
| | | | | | | | Original commit message from CVS: Patch by: Håvard Graff <havard dot graff at tandberg dot com> * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Fix debug by logging the right seqnum.
* gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map ↵Olivier Crete2009-08-111-0/+13
| | | | | | | | | | signal. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/gstrtpbin.c: (get_pt_map): Release lock before emitting the request-pt-map signal. Fixes #543480.
* gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).Peter Kjellerstedt2009-08-112-3/+3
| | | | | | | | Original commit message from CVS: * ChangeLog: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr): Corrected a typo (interpollate -> interpolate).
* gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam ↵Peter Kjellerstedt2009-08-113-19/+18
| | | | | | | | | | | | | | | | when a pipeline is running normally. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_send_rtp): * gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp): Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
* gst/rtpmanager/: Do not mix the use of g_get_current_time() with ↵Peter Kjellerstedt2009-08-113-91/+85
| | | | | | | | | | | | | | | | | | | 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().
* Final round of doc updates.Stefan Kost2009-08-111-1/+2
| | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/speed/gstspeed.c: * gst/speexresample/gstspeexresample.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/dvb/gstdvbsrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/wininet/gstwininetsrc.c: Final round of doc updates.
* gst/: More doc updates. More xrefs.Stefan Kost2009-08-116-153/+100
| | | | | | | | | | | | | Original commit message from CVS: * gst/deinterlace/gstdeinterlace.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/sdp/gstsdpdemux.c: More doc updates. More xrefs.
* Do not use short_description in section docs for elements. We extract them ↵Stefan Kost2009-08-116-7/+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.
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