sess->ptmap = g_hash_table_new (NULL, NULL);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
+ /* set NTP base or new session */
+ g_object_set (session, "ntp-ns-base", rtpbin->priv->ntp_ns_base, NULL);
+
/* provide clock_rate to the session manager when needed */
g_signal_connect (session, "request-pt-map",
(GCallback) pt_map_requested, sess);
priv->clock_base = exttimestamp;
}
- /* take rtp timestamp offset into account, this can wrap around */
+ /* take rtp timestamp offset into account, this should not wrap around since
+ * we are dealing with the extended timestamp here. */
exttimestamp -= priv->clock_base;
/* bring timestamp to gst time */
/* store this so that we can safely sync on the peer buffers. */
JBUF_LOCK (priv);
priv->peer_latency = min_latency;
- JBUF_UNLOCK (priv);
-
our_latency = ((guint64) priv->latency_ms) * GST_MSECOND;
+ JBUF_UNLOCK (priv);
GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
GST_TIME_ARGS (our_latency));
{
guint new_latency, old_latency;
- /* FIXME, not threadsafe */
new_latency = g_value_get_uint (value);
- old_latency = priv->latency_ms;
+ JBUF_LOCK (priv);
+ old_latency = priv->latency_ms;
priv->latency_ms = new_latency;
+ JBUF_UNLOCK (priv);
/* post message if latency changed, this will inform the parent pipeline
* that a latency reconfiguration is possible/needed. */
switch (prop_id) {
case PROP_LATENCY:
+ JBUF_LOCK (priv);
g_value_set_uint (value, priv->latency_ms);
+ JBUF_UNLOCK (priv);
break;
case PROP_DROP_ON_LATENCY:
g_value_set_boolean (value, priv->drop_on_latency);
return ret;
}
+static GstCaps *
+gst_rtp_session_getcaps_send_rtp (GstPad * pad)
+{
+ GstRtpSession *rtpsession;
+ GstRtpSessionPrivate *priv;
+ GstCaps *result;
+ GstStructure *s1, *s2;
+
+ rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
+ priv = rtpsession->priv;
+
+ /* we can basically accept anything but we prefer to receive packets with our
+ * internal SSRC so that we don't have to patch it. Create a structure with
+ * the SSRC and another one without. */
+ s1 = gst_structure_new ("application/x-rtp",
+ "ssrc", G_TYPE_UINT, priv->session->source->ssrc, NULL);
+ s2 = gst_structure_new ("application/x-rtp", NULL);
+
+ result = gst_caps_new_full (s1, s2, NULL);
+
+ GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
+
+ gst_object_unref (rtpsession);
+
+ return result;
+}
+
/* Recieve an RTP packet to be send to the receivers, send to RTP session
* manager and forward to send_rtp_src.
*/
timestamp);
/* convert to NTP time by adding the NTP base */
ntpnstime += priv->ntpnsbase;
- } else
+ } else {
+ /* no timestamp, we could take the current running_time and convert it to
+ * NTP time. */
ntpnstime = -1;
+ }
ret = rtp_session_send_rtp (priv->session, buffer, ntpnstime);
"send_rtp_sink");
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
+ gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
+ gst_rtp_session_getcaps_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
gst_rtp_session_event_send_rtp_sink);
gst_pad_set_internal_link_function (rtpsession->send_rtp_sink,