From a35d1dde421be0655eb36fed9f415a25f5fa00e0 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 5 Sep 2008 13:52:34 +0000 Subject: 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. --- gst/rtpmanager/rtpsession.c | 36 ++++++++++++++++++++++++++++-------- 1 file changed, 28 insertions(+), 8 deletions(-) (limited to 'gst/rtpmanager/rtpsession.c') diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 947fef7e..428181f2 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -40,6 +40,7 @@ enum SIGNAL_ON_BYE_SSRC, SIGNAL_ON_BYE_TIMEOUT, SIGNAL_ON_TIMEOUT, + SIGNAL_ON_SENDER_TIMEOUT, LAST_SIGNAL }; @@ -212,6 +213,18 @@ rtp_session_class_init (RTPSessionClass * klass) G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); + /** + * RTPSession::on-sender-timeout: + * @session: the object which received the signal + * @src: the RTPSource that timed out + * + * Notify of an SSRC that was a sender but timed out and became a receiver. + */ + rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] = + g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout), + NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, + RTP_TYPE_SOURCE); g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE, g_param_spec_object ("internal-source", "Internal Source", @@ -513,6 +526,15 @@ on_timeout (RTPSession * sess, RTPSource * source) RTP_SESSION_LOCK (sess); } +static void +on_sender_timeout (RTPSession * sess, RTPSource * source) +{ + RTP_SESSION_UNLOCK (sess); + g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, + source); + RTP_SESSION_LOCK (sess); +} + /** * rtp_session_new: * @@ -908,9 +930,8 @@ check_collision (RTPSession * sess, RTPSource * source, RTPArrivalStats * arrival, gboolean rtp) { /* If we have not arrival address, we can't do collision checking */ - if (!arrival->have_address) { + if (!arrival->have_address) return FALSE; - } if (sess->source != source) { /* This is not our local source, but lets check if two remote @@ -1479,12 +1500,6 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet, if (!source) return; - /* we somehow need to transfer the clock_base and the base time to the next - * element, we use the offset and offset_end fields in the buffer for this - * hack */ - GST_BUFFER_OFFSET (packet->buffer) = source->clock_base; - GST_BUFFER_OFFSET_END (packet->buffer) = source->clock_base_time; - prevsender = RTP_SOURCE_IS_SENDER (source); /* first update the source */ @@ -2096,6 +2111,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data) { gboolean remove = FALSE; gboolean byetimeout = FALSE; + gboolean sendertimeout = FALSE; gboolean is_sender, is_active; RTPSession *sess = data->sess; GstClockTime interval; @@ -2138,6 +2154,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data) GST_TIME_ARGS (source->last_rtp_activity)); source->is_sender = FALSE; sess->stats.sender_sources--; + sendertimeout = TRUE; } } } @@ -2153,6 +2170,9 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data) on_bye_timeout (sess, source); else on_timeout (sess, source); + } else { + if (sendertimeout) + on_sender_timeout (sess, source); } return remove; } -- cgit v1.2.1