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a35d1dde42
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. |
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| aiffparse | ||
| app | ||
| audioresample | ||
| bayer | ||
| cdxaparse | ||
| chart | ||
| colorspace | ||
| dccp | ||
| deinterlace | ||
| deinterlace2 | ||
| dvdspu | ||
| festival | ||
| filter | ||
| flv | ||
| freeze | ||
| games | ||
| h264parse | ||
| librfb | ||
| mixmatrix | ||
| modplug | ||
| mpeg1sys | ||
| mpeg4videoparse | ||
| mpegdemux | ||
| mpegtsmux | ||
| mpegtsparse | ||
| mpegvideoparse | ||
| mve | ||
| nsf | ||
| nuvdemux | ||
| overlay | ||
| passthrough | ||
| pcapparse | ||
| playondemand | ||
| rawparse | ||
| real | ||
| rtjpeg | ||
| rtpmanager | ||
| scaletempo | ||
| sdp | ||
| selector | ||
| smooth | ||
| smoothwave | ||
| speed | ||
| speexresample | ||
| stereo | ||
| subenc | ||
| tta | ||
| vbidec | ||
| videodrop | ||
| videosignal | ||
| virtualdub | ||
| vmnc | ||
| y4m | ||
| Makefile.am | ||