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rtp: include downstream latency in SR calculations

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When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
This commit is contained in:
Wim Taymans 2013-01-07 15:45:10 +01:00
parent c631ed3300
commit 87f7d6b9bf
1 changed files with 32 additions and 0 deletions
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32
gst/rtpmanager/gstrtpsession.c
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@ -242,6 +242,8 @@ struct _GstRtpSessionPrivate
/* caps mapping */ /* caps mapping */
GHashTable *ptmap; GHashTable *ptmap;
GstClockTime send_latency;
gboolean use_pipeline_clock; gboolean use_pipeline_clock;
}; };
@ -1712,6 +1714,33 @@ gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
return ret; return ret;
} }
static gboolean
gst_rtp_session_event_send_rtp_src (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
/* save the latency, we need this to know when an RTP packet will be
* rendered by the sink */
gst_event_parse_latency (event, &rtpsession->priv->send_latency);
ret = gst_pad_event_default (pad, parent, event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static GstCaps * static GstCaps *
gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession, gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
GstCaps * filter) GstCaps * filter)
@ -1829,6 +1858,7 @@ gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
running_time = running_time =
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
timestamp); timestamp);
running_time += priv->send_latency;
} else { } else {
/* no timestamp. */ /* no timestamp. */
running_time = -1; running_time = -1;
@ -2011,6 +2041,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession)
"send_rtp_src"); "send_rtp_src");
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src, gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src,
gst_rtp_session_iterate_internal_links); gst_rtp_session_iterate_internal_links);
gst_pad_set_event_function (rtpsession->send_rtp_src,
gst_rtp_session_event_send_rtp_src);
gst_pad_set_active (rtpsession->send_rtp_src, TRUE); gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);