[farsight2/master] Implement the callback after a timeout if there is no rtcp

[Olivier Crête]

---
 gst/fsrtpconference/fs-rtp-session.c   |   50 ++++++++++++++++++++++++++++++++
 gst/fsrtpconference/fs-rtp-session.h   |    3 ++
 gst/fsrtpconference/fs-rtp-substream.c |   31 +++++++++++++++++++
 3 files changed, 84 insertions(+), 0 deletions(-)

diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c
index 54ebdb2..3fb82c2 100644
--- a/gst/fsrtpconference/fs-rtp-session.c
+++ b/gst/fsrtpconference/fs-rtp-session.c
@@ -1712,6 +1712,12 @@ fs_rtp_session_new_recv_pad (FsRtpSession *session, GstPad *new_pad,
   FS_RTP_SESSION_LOCK (session);
   stream = fs_rtp_session_get_stream_by_ssrc (session, ssrc);
 
+  /* Add the substream directly if the no_rtcp_timeout is 0 */
+  if (!stream &&
+      no_rtcp_timeout == 0 &&
+      g_list_length (session->priv->streams) == 1)
+    stream = g_object_ref (g_list_first (session->priv->streams)->data);
+
   if (!stream)
     session->priv->free_substreams =
       g_list_prepend (session->priv->free_substreams, substream);
@@ -2434,3 +2440,47 @@ fs_rtp_session_associate_ssrc_cname (FsRtpSession *session,
         error->message);
   g_clear_error (&error);
 }
+
+void
+fs_rtp_session_substream_timedout (FsRtpSession *session,
+    gpointer substream)
+{
+  GError *error = NULL;
+  FsRtpSubStream *realsubstream = substream;
+  FS_RTP_SESSION_LOCK (session);
+
+  if (g_list_length (session->priv->streams) != 1)
+  {
+    guint ssrc, pt;
+    gint timeout;
+    g_object_get (realsubstream,
+        "ssrc", &ssrc,
+        "pt", &pt,
+        "no-rtcp-timeout", &timeout,
+        NULL);
+    GST_WARNING ("The substream for SSRC %x and pt %u did not receive RTCP"
+        " for %d milliseconds, but we have more than one stream so we can"
+        " not associate it.", ssrc, pt, timeout);
+    goto done;
+  }
+
+  session->priv->free_substreams =
+    g_list_remove (session->priv->free_substreams,
+        realsubstream);
+
+  if (!fs_rtp_stream_add_substream (
+          g_list_first (session->priv->streams)->data,
+          realsubstream, &error))
+  {
+    fs_session_emit_error (FS_SESSION (session),
+        error ? error->code : FS_ERROR_INTERNAL,
+        "Could not link the substream to a stream",
+        error ? error->message : "No error message");
+  }
+  g_clear_error (&error);
+
+
+ done:
+
+  FS_RTP_SESSION_UNLOCK (session);
+}
diff --git a/gst/fsrtpconference/fs-rtp-session.h b/gst/fsrtpconference/fs-rtp-session.h
index 7445b2b..575c095 100644
--- a/gst/fsrtpconference/fs-rtp-session.h
+++ b/gst/fsrtpconference/fs-rtp-session.h
@@ -109,6 +109,9 @@ void fs_rtp_session_associate_ssrc_cname (FsRtpSession *session,
     const gchar *cname);
 
 
+void fs_rtp_session_substream_timedout (FsRtpSession *session,
+    gpointer substream);
+
 G_END_DECLS
 
 #endif /* __FS_RTP_SESSION_H__ */
diff --git a/gst/fsrtpconference/fs-rtp-substream.c b/gst/fsrtpconference/fs-rtp-substream.c
index 48fe2b4..5fcec23 100644
--- a/gst/fsrtpconference/fs-rtp-substream.c
+++ b/gst/fsrtpconference/fs-rtp-substream.c
@@ -237,6 +237,23 @@ fs_rtp_sub_stream_init (FsRtpSubStream *self)
   self->priv->receiving = TRUE;
 }
 
+static gboolean
+_no_rtcp_timeout (gpointer user_data)
+{
+  FsRtpSubStream *self = FS_RTP_SUB_STREAM (user_data);
+
+  FS_RTP_SESSION_LOCK (self->priv->session);
+
+  if (!self->priv->stream)
+    fs_rtp_session_substream_timedout (self->priv->session, self);
+
+  if (self->priv->no_rtcp_timeout_id)
+    self->priv->no_rtcp_timeout_id = 0;
+
+  FS_RTP_SESSION_UNLOCK (self->priv->session);
+
+  return FALSE;
+}
 
 static void
 fs_rtp_sub_stream_constructed (GObject *object)
@@ -279,6 +296,10 @@ fs_rtp_sub_stream_constructed (GObject *object)
       self->priv->ssrc, self->priv->pt);
     return;
   }
+
+  if (self->priv->no_rtcp_timeout > 0)
+    self->priv->no_rtcp_timeout_id = g_timeout_add (self->priv->no_rtcp_timeout,
+        _no_rtcp_timeout, self);
 }
 
 
@@ -290,6 +311,16 @@ fs_rtp_sub_stream_dispose (GObject *object)
   if (self->priv->disposed)
     return;
 
+
+  FS_RTP_SESSION_LOCK (self->priv->session);
+  if (self->priv->no_rtcp_timeout_id)
+  {
+    g_source_remove (self->priv->no_rtcp_timeout_id);
+    self->priv->no_rtcp_timeout_id = 0;
+  }
+  FS_RTP_SESSION_UNLOCK (self->priv->session);
+
+
   if (self->priv->output_ghostpad) {
     gst_element_remove_pad (GST_ELEMENT (self->priv->conference),
       self->priv->output_ghostpad);
-- 
1.5.6.5