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authorWim Taymans <wim.taymans@gmail.com>2008-09-05 13:52:34 +0000
committerWim Taymans <wim.taymans@gmail.com>2008-09-05 13:52:34 +0000
commita35d1dde421be0655eb36fed9f415a25f5fa00e0 (patch)
treec0bfaa3e8fccfad821f2175dc419987caf0c2636 /gst/rtpmanager/gstrtpjitterbuffer.h
parent64cd01e7e8a143e523466c911f7bb2e148508c3b (diff)
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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.
Diffstat (limited to 'gst/rtpmanager/gstrtpjitterbuffer.h')
-rw-r--r--gst/rtpmanager/gstrtpjitterbuffer.h3
1 files changed, 3 insertions, 0 deletions
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.h b/gst/rtpmanager/gstrtpjitterbuffer.h
index 290aee09..15185a25 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.h
+++ b/gst/rtpmanager/gstrtpjitterbuffer.h
@@ -79,6 +79,9 @@ struct _GstRtpJitterBufferClass
GType gst_rtp_jitter_buffer_get_type (void);
+void gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer *buffer,
+ guint64 *rtptime, guint64 *timestamp);
+
G_END_DECLS
#endif /* __GST_RTP_JITTER_BUFFER_H__ */