gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.

Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_send_rtp):
* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp):
Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
pipeline is running normally.

4 files changed, 31 insertions, 19 deletions

diff --git a/ChangeLog b/ChangeLog
index 8f9765c4..f463c12d 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,18 @@
 2008-07-03  Peter Kjellerstedt  <pkj@axis.com>
 
+	* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
+	(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
+	(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
+	(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
+	* gst/rtpmanager/rtpsession.c: (source_push_rtp),
+	(rtp_session_send_rtp):
+	* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
+	(rtp_source_process_rtp), (rtp_source_send_rtp):
+	Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
+	pipeline is running normally.
+
+2008-07-03  Peter Kjellerstedt  <pkj@axis.com>
+
 	* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
 	(gst_rtp_session_finalize), (rtcp_thread),
 	(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index a3359e45..b96a1dfc 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -1062,7 +1062,7 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
   priv = rtpsession->priv;
 
   if (rtpsession->recv_rtp_src) {
-    GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet");
+    GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
     result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
   } else {
     GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
@@ -1085,7 +1085,7 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
   rtpsession = GST_RTP_SESSION (user_data);
   priv = rtpsession->priv;
 
-  GST_DEBUG_OBJECT (rtpsession, "sending RTP packet");
+  GST_LOG_OBJECT (rtpsession, "sending RTP packet");
 
   if (rtpsession->send_rtp_src) {
     result = gst_pad_push (rtpsession->send_rtp_src, buffer);
@@ -1119,7 +1119,7 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
       gst_caps_unref (caps);
     }
     gst_buffer_set_caps (buffer, caps);
-    GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
+    GST_LOG_OBJECT (rtpsession, "sending RTCP");
     result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
   } else {
     GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
@@ -1152,7 +1152,7 @@ gst_rtp_session_sync_rtcp (RTPSession * sess,
       gst_caps_unref (caps);
     }
     gst_buffer_set_caps (buffer, caps);
-    GST_DEBUG_OBJECT (rtpsession, "sending Sync RTCP");
+    GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
     result = gst_pad_push (rtpsession->sync_src, buffer);
   } else {
     GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
@@ -1390,7 +1390,7 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
   rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
   priv = rtpsession->priv;
 
-  GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
+  GST_LOG_OBJECT (rtpsession, "received RTP packet");
 
   /* get NTP time when this packet was captured, this depends on the timestamp. */
   timestamp = GST_BUFFER_TIMESTAMP (buffer);
@@ -1467,7 +1467,7 @@ gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
   rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
   priv = rtpsession->priv;
 
-  GST_DEBUG_OBJECT (rtpsession, "received RTCP packet");
+  GST_LOG_OBJECT (rtpsession, "received RTCP packet");
 
   current_time = gst_clock_get_time (priv->sysclock);
   ret = rtp_session_process_rtcp (priv->session, buffer, current_time);
@@ -1614,7 +1614,7 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
   rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
   priv = rtpsession->priv;
 
-  GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
+  GST_LOG_OBJECT (rtpsession, "received RTP packet");
 
   /* get NTP time when this packet was captured, this depends on the timestamp. */
   timestamp = GST_BUFFER_TIMESTAMP (buffer);
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index c7ab410f..59b1d8d0 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -812,7 +812,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
   GstFlowReturn result = GST_FLOW_OK;
 
   if (source == session->source) {
-    GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
+    GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
 
     RTP_SESSION_UNLOCK (session);
 
@@ -824,7 +824,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
       gst_buffer_unref (buffer);
 
   } else {
-    GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
+    GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
     RTP_SESSION_UNLOCK (session);
 
     if (session->callbacks.process_rtp)
@@ -1772,7 +1772,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
   if (!gst_rtp_buffer_validate (buffer))
     goto invalid_packet;
 
-  GST_DEBUG ("received RTP packet for sending");
+  GST_LOG ("received RTP packet for sending");
 
   RTP_SESSION_LOCK (sess);
   source = sess->source;
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 7eab91e0..50170d10 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -708,13 +708,13 @@ push_packet (RTPSource * src, GstBuffer * buffer)
   while (!g_queue_is_empty (src->packets)) {
     GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
 
-    GST_DEBUG ("pushing queued packet");
+    GST_LOG ("pushing queued packet");
     if (src->callbacks.push_rtp)
       src->callbacks.push_rtp (src, buffer, src->user_data);
     else
       gst_buffer_unref (buffer);
   }
-  GST_DEBUG ("pushing new packet");
+  GST_LOG ("pushing new packet");
   /* push packet */
   if (src->callbacks.push_rtp)
     ret = src->callbacks.push_rtp (src, buffer, src->user_data);
@@ -763,7 +763,7 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
 
   pt = gst_rtp_buffer_get_payload_type (buffer);
 
-  GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
+  GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
 
   /* get clockrate */
   if ((clock_rate = get_clock_rate (src, pt)) == -1)
@@ -802,7 +802,7 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
   src->stats.prev_rtptime = src->stats.last_rtptime;
   src->stats.last_rtptime = rtparrival;
 
-  GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
+  GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
       rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
 
   return;
@@ -937,7 +937,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
   src->is_sender = TRUE;
   src->validated = TRUE;
 
-  GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
+  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 */
@@ -1018,7 +1018,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
   ext_rtptime = src->last_rtptime;
   ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
 
-  GST_DEBUG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
+  GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
       src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
 
   if (ext_rtptime > src->last_rtptime) {
@@ -1028,7 +1028,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
     /* calc the diff so we can detect drift at the sender. This can also be used
      * to guestimate the clock rate if the NTP time is locked to the RTP
      * timestamps (as is the case when the capture device is providing the clock). */
-    GST_DEBUG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
+    GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
         GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
   }
 
@@ -1053,8 +1053,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
           src->ssrc);
       gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
     }
-    GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
-        src->stats.packets_sent);
+    GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
     result = src->callbacks.push_rtp (src, buffer, src->user_data);
   } else {
     GST_WARNING ("no callback installed, dropping packet");