gst/rtpmanager/gstrtpsession.c: Remove debug.

Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
Remove debug.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_sdes), (calculate_rtcp_interval),
(rtp_session_next_timeout), (session_report_blocks):
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
Improve debugging
Fix interval for BYE/RTCP packets.

4 files changed, 26 insertions, 11 deletions

diff --git a/ChangeLog b/ChangeLog
index 0bc7ba43..4ebf8d7b 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,15 @@
+2007-04-29  Wim Taymans  <wim@fluendo.com>
+
+	* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
+	Remove debug.
+
+	* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
+	(rtp_session_process_sdes), (calculate_rtcp_interval),
+	(rtp_session_next_timeout), (session_report_blocks):
+	* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
+	Improve debugging
+	Fix interval for BYE/RTCP packets.
+
 2007-04-29  Thomas Vander Stichele  <thomas at apestaart dot org>
 
 	* docs/plugins/gst-plugins-bad-plugins.args:
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 9545a92d..12a46fe6 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -497,8 +497,6 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
 
   GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
 
-  gst_util_dump_mem (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
-
   if (rtpsession->send_rtcp_src) {
     result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
   } else {
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 7244f5fb..af418abf 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -1051,7 +1051,8 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
   gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
       &packet_count, &octet_count);
 
-  GST_DEBUG ("got SR packet: SSRC %08x", senderssrc);
+  GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
+      senderssrc, GST_TIME_ARGS (arrival->time));
 
   source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
 
@@ -1158,7 +1159,8 @@ rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
 
       gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
 
-      GST_DEBUG ("entry %d, type %d, len %d, data %s", j, type, len, data);
+      GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
+          data);
 
       more_entries = gst_rtcp_packet_sdes_next_entry (packet);
       j++;
@@ -1395,14 +1397,14 @@ calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
   GstClockTime result;
 
   if (sess->source->received_bye) {
+    result = rtp_stats_calculate_bye_interval (&sess->stats);
+  } else {
     result = rtp_stats_calculate_rtcp_interval (&sess->stats,
         RTP_SOURCE_IS_SENDER (sess->source), first);
-  } else {
-    result = rtp_stats_calculate_bye_interval (&sess->stats);
   }
 
-  GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (result));
+  GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
+      GST_TIME_ARGS (result), first);
 
   if (!deterministic)
     result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
@@ -1495,7 +1497,7 @@ rtp_session_next_timeout (RTPSession * sess, GstClockTime time)
       result = GST_CLOCK_TIME_NONE;
     else if (sess->stats.active_sources >= 50)
       /* reconsider BYE if members >= 50 */
-      result = time + calculate_rtcp_interval (sess, FALSE, TRUE);;
+      result = time + calculate_rtcp_interval (sess, FALSE, TRUE);
   } else {
     if (sess->first_rtcp)
       /* we are called for the first time */
@@ -1597,10 +1599,14 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
           extended_max, stats->jitter >> 4);
 
       if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {
+        GstClockTime diff;
+
         /* LSR is middle bits of the last ntptime */
         LSR = (ntptime >> 16) & 0xffffffff;
+        diff = data->time - time;
+        GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
         /* DLSR, delay since last SR is expressed in 1/65536 second units */
-        DLSR = gst_util_uint64_scale_int (data->time - time, 65536, GST_SECOND);
+        DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
       } else {
         /* No valid SR received, LSR/DLSR are set to 0 then */
         LSR = 0;
diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
index 1e18f45e..55cf2e05 100644
--- a/gst/rtpmanager/rtpstats.c
+++ b/gst/rtpmanager/rtpstats.c
@@ -56,7 +56,6 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
   gdouble interval;
   gdouble rtcp_min_time;
 
-
   /* Very first call at application start-up uses half the min
    * delay for quicker notification while still allowing some time
    * before reporting for randomization and to learn about other