rtpsource: refactor receiver stats update

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index dea2e78..1c8f7f8 100644 (file)
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -977,26 +977,13 @@ do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
   }
 }
 
-/**
- * rtp_source_process_rtp:
- * @src: an #RTPSource
- * @pinfo: an #RTPPacketInfo
- *
- * Let @src handle the incomming RTP packet described in @pinfo.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
+static gboolean
+update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo)
 {
-  GstFlowReturn result = GST_FLOW_OK;
   guint16 seqnr, udelta;
   RTPSourceStats *stats;
   guint16 expected;
 
-  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
-  g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
-
   stats = &src->stats;
 
   seqnr = pinfo->seqnum;
@@ -1071,41 +1058,68 @@ rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
   src->stats.packets_received++;
   /* for the bitrate estimation */
   src->bytes_received += pinfo->payload_len;
-  /* the source that sent the packet must be a sender */
-  src->is_sender = TRUE;
-  src->validated = TRUE;
-
-  do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
 
   GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
       seqnr, src->stats.packets_received, src->stats.octets_received);
 
-  /* calculate jitter for the stats */
-  calculate_jitter (src, pinfo);
-
-  /* we're ready to push the RTP packet now */
-  result = push_packet (src, pinfo->data);
-  pinfo->data = NULL;
-
-done:
-  return result;
+  return TRUE;
 
   /* ERRORS */
+done:
+  {
+    return FALSE;
+  }
 bad_sequence:
   {
     GST_WARNING ("unacceptable seqnum received");
-    return GST_FLOW_OK;
+    return FALSE;
   }
 probation_seqnum:
   {
     GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
     src->curr_probation = src->probation;
     src->stats.max_seq = seqnr;
-    return GST_FLOW_OK;
+    return FALSE;
   }
 }
 
 /**
+ * rtp_source_process_rtp:
+ * @src: an #RTPSource
+ * @pinfo: an #RTPPacketInfo
+ *
+ * Let @src handle the incomming RTP packet described in @pinfo.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+GstFlowReturn
+rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
+{
+  GstFlowReturn result;
+
+  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
+  g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
+
+  if (!update_receiver_stats (src, pinfo))
+    return GST_FLOW_OK;
+
+  /* the source that sent the packet must be a sender */
+  src->is_sender = TRUE;
+  src->validated = TRUE;
+
+  do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
+
+  /* calculate jitter for the stats */
+  calculate_jitter (src, pinfo);
+
+  /* we're ready to push the RTP packet now */
+  result = push_packet (src, pinfo->data);
+  pinfo->data = NULL;
+
+  return result;
+}
+
+/**
  * rtp_source_mark_bye:
  * @src: an #RTPSource
  * @reason: the reason for leaving