gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes...
authorWim Taymans <wim.taymans@gmail.com>
Fri, 5 Sep 2008 13:52:34 +0000 (13:52 +0000)
committerTim-Philipp Müller <tim.muller@collabora.co.uk>
Tue, 11 Aug 2009 01:30:37 +0000 (02:30 +0100)
commit85e26f65468b6407ef753220c70695ef87700045
treee29d4d0f0f987f8021e2f3cba549a563f6fddd13
parent5c89bb2ab3a15eafcb59581cfc2bf5e477cafc73
gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender 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.
13 files changed:
gst/rtpmanager/gstrtpbin.c
gst/rtpmanager/gstrtpbin.h
gst/rtpmanager/gstrtpjitterbuffer.c
gst/rtpmanager/gstrtpjitterbuffer.h
gst/rtpmanager/gstrtpsession.c
gst/rtpmanager/gstrtpsession.h
gst/rtpmanager/rtpjitterbuffer.c
gst/rtpmanager/rtpjitterbuffer.h
gst/rtpmanager/rtpsession.c
gst/rtpmanager/rtpsession.h
gst/rtpmanager/rtpsource.c
gst/rtpmanager/rtpsource.h
gst/rtpmanager/rtpstats.h