gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 6f288bd..eb028fb 100644 (file)
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -427,6 +427,9 @@ create_session (GstRtpBin * rtpbin, gint id)
   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);

diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 497ce89..327ff0a 100644 (file)
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -1018,7 +1018,8 @@ again:
       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 */
@@ -1218,9 +1219,8 @@ gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query)
           /* 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));
@@ -1263,11 +1263,12 @@ gst_rtp_jitter_buffer_set_property (GObject * object,
     {
       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. */
@@ -1306,7 +1307,9 @@ gst_rtp_jitter_buffer_get_property (GObject * object,
 
   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);

diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 44f6535..6d4cf8b 100644 (file)
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -1226,6 +1226,33 @@ gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event)
   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.
  */
@@ -1252,8 +1279,11 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
         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);
 
@@ -1341,6 +1371,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession)
       "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,