gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qt...

Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).

diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 9f5d58d842491bfbbd5d04dba971dbd8bbc79912..749079123673c02e57d6a1f899bcc2a65723712c 100644 (file)
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -364,7 +364,7 @@ rtp_session_get_bandwidth (RTPSession * sess)
  * @bandwidth: the RTCP bandwidth
  *
  * Set the bandwidth that should be used for RTCP
- * messages. 
+ * messages.
  */
 void
 rtp_session_set_rtcp_bandwidth (RTPSession * sess, gdouble bandwidth)
@@ -395,7 +395,7 @@ rtp_session_get_rtcp_bandwidth (RTPSession * sess)
  * @sess: an #RTPSession
  * @cname: a CNAME for the session
  *
- * Set the CNAME for the session. 
+ * Set the CNAME for the session.
  */
 void
 rtp_session_set_cname (RTPSession * sess, const gchar * cname)
@@ -427,7 +427,7 @@ rtp_session_get_cname (RTPSession * sess)
  * @sess: an #RTPSession
  * @name: a NAME for the session
  *
- * Set the NAME for the session. 
+ * Set the NAME for the session.
  */
 void
 rtp_session_set_name (RTPSession * sess, const gchar * name)
@@ -459,7 +459,7 @@ rtp_session_get_name (RTPSession * sess)
  * @sess: an #RTPSession
  * @email: an EMAIL for the session
  *
- * Set the EMAIL the session. 
+ * Set the EMAIL the session.
  */
 void
 rtp_session_set_email (RTPSession * sess, const gchar * email)
@@ -491,7 +491,7 @@ rtp_session_get_email (RTPSession * sess)
  * @sess: an #RTPSession
  * @phone: a PHONE for the session
  *
- * Set the PHONE the session. 
+ * Set the PHONE the session.
  */
 void
 rtp_session_set_phone (RTPSession * sess, const gchar * phone)
@@ -523,7 +523,7 @@ rtp_session_get_phone (RTPSession * sess)
  * @sess: an #RTPSession
  * @location: a LOCATION for the session
  *
- * Set the LOCATION the session. 
+ * Set the LOCATION the session.
  */
 void
 rtp_session_set_location (RTPSession * sess, const gchar * location)
@@ -555,7 +555,7 @@ rtp_session_get_location (RTPSession * sess)
  * @sess: an #RTPSession
  * @tool: a TOOL for the session
  *
- * Set the TOOL the session. 
+ * Set the TOOL the session.
  */
 void
 rtp_session_set_tool (RTPSession * sess, const gchar * tool)
@@ -587,7 +587,7 @@ rtp_session_get_tool (RTPSession * sess)
  * @sess: an #RTPSession
  * @note: a NOTE for the session
  *
- * Set the NOTE the session. 
+ * Set the NOTE the session.
  */
 void
 rtp_session_set_note (RTPSession * sess, const gchar * note)
@@ -1228,7 +1228,7 @@ rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
     members = sess->stats.active_sources;
 
     if (!sess->source->received_bye && members < pmembers) {
-      /* some members went away since the previous timeout estimate. 
+      /* some members went away since the previous timeout estimate.
        * Perform reverse reconsideration but only when we are not scheduling a
        * BYE ourselves. */
       if (arrival->time < sess->next_rtcp_check_time) {
@@ -1612,7 +1612,8 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
       extended_max = stats->cycles + stats->max_seq;
       expected = extended_max - stats->base_seq + 1;
 
-      GST_DEBUG ("ext_max %d, expected %d, received %d, base_seq %d",
+      GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
+          ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
           extended_max, expected, stats->packets_received, stats->base_seq);
 
       lost = expected - stats->packets_received;
@@ -1632,7 +1633,8 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
 
       GST_DEBUG ("add RR for SSRC %08x", source->ssrc);
       /* we scaled the jitter up for additional precision */
-      GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
+      GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
+          ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
           extended_max, stats->jitter >> 4);
 
       if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 7af7467168f6a693f600dffd4b8f3747607e76fa..8007d54b5dae3ab46f950a1dd4d42840e0b50061 100644 (file)
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -347,7 +347,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
       src->probation--;
       src->stats.max_seq = seqnr;
       if (src->probation == 0) {
-        GST_DEBUG ("probation done!", src->probation);
+        GST_DEBUG ("probation done!");
         init_seq (src, seqnr);
       } else {
         GstBuffer *q;
@@ -470,7 +470,8 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
 
   /* push packet */
   if (src->callbacks.push_rtp) {
-    GST_DEBUG ("pushing RTP packet %u", src->stats.packets_sent);
+    GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
+        src->stats.packets_sent);
     result = src->callbacks.push_rtp (src, buffer, src->user_data);
   } else {
     GST_DEBUG ("no callback installed");
@@ -500,9 +501,10 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
 
   g_return_if_fail (RTP_IS_SOURCE (src));
 
-  GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %u, PC %u, OC %u",
-      src->ssrc, ntptime >> 32, ntptime & 0xffffffff, rtptime, packet_count,
-      octet_count);
+  GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
+      ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
+      (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
+      packet_count, octet_count);
 
   curridx = src->stats.curr_sr ^ 1;
   curr = &src->stats.sr[curridx];
@@ -543,9 +545,10 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
 
   g_return_if_fail (RTP_IS_SOURCE (src));
 
-  GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
-      ", PL %u, HS %u, JITTER %u, LSR %08x, DLSR %08x", src->ssrc, fractionlost,
-      packetslost, exthighestseq, jitter, lsr, dlsr);
+  GST_DEBUG ("got RB packet: SSRC %08x, FL %" G_GUINT32_FORMAT ""
+      ", PL %d, HS %" G_GUINT32_FORMAT ", JITTER %" G_GUINT32_FORMAT
+      ", LSR %08x, DLSR %08x", src->ssrc, fractionlost, packetslost,
+      exthighestseq, jitter, lsr, dlsr);
 
   curridx = src->stats.curr_rr ^ 1;
   curr = &src->stats.rr[curridx];