gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...

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>).

5 files changed, 60 insertions, 30 deletions

diff --git a/ChangeLog b/ChangeLog
index fb03cc33..a5a9594d 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,24 @@
 2007-05-10  Stefan Kost  <ensonic@users.sf.net>
 
+	* 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>).
+
+2007-05-10  Stefan Kost  <ensonic@users.sf.net>
+
 	* gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
 	  async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
 	  async_jitter_queue_set_low_threshold,
@@ -11,7 +30,7 @@
 	  async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
 	  async_jitter_queue_set_flushing_unlocked,
 	  async_jitter_queue_unset_flushing_unlocked):
-	  Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
+	  Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>).
 
 2007-05-10  Stefan Kost  <ensonic@users.sf.net>
 
diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c
index 66bc04ea..ec438d5c 100644
--- a/gst/qtdemux/qtdemux.c
+++ b/gst/qtdemux/qtdemux.c
@@ -585,7 +585,7 @@ gst_qtdemux_move_stream (GstQTDemux * qtdemux, QtDemuxStream * str,
  * streaming from the desired position.
  *
  * Keyframe seeking is a little more complicated when dealing with
- * segments. Ideally we want to move to the previous keyframe in 
+ * segments. Ideally we want to move to the previous keyframe in
  * the segment but there might not be a keyframe in the segment. In
  * fact, none of the segments could contain a keyframe. We take a
  * practical approach: seek to the previous keyframe in the segment,
@@ -1024,7 +1024,7 @@ beach:
  * @offset is an absolute global position over all the segments.
  *
  * This will push out a NEWSEGMENT event with the right values and
- * position the stream index to the first decodable sample before 
+ * position the stream index to the first decodable sample before
  * @offset.
  */
 static gboolean
@@ -1107,7 +1107,7 @@ gst_qtdemux_activate_segment (GstQTDemux * qtdemux, QtDemuxStream * stream,
 }
 
 /* prepare to get the current sample of @stream, getting essential values.
- * 
+ *
  * This function will also prepare and send the segment when needed.
  *
  * Return FALSE if the stream is EOS.
@@ -1142,6 +1142,9 @@ gst_qtdemux_prepare_current_sample (GstQTDemux * qtdemux,
   if (stream->segment_index != seg_idx)
     gst_qtdemux_activate_segment (qtdemux, stream, seg_idx, time_position);
 
+  GST_LOG_OBJECT (qtdemux, "segment active, index = %lu of %lu",
+      stream->sample_index, stream->n_samples);
+
   if (stream->sample_index >= stream->n_samples)
     goto eos;
 
@@ -1248,6 +1251,7 @@ gst_qtdemux_combine_flows (GstQTDemux * demux, QtDemuxStream * stream,
   /* if we get here, all other pads were unlinked and we return
    * NOT_LINKED then */
 done:
+  GST_LOG_OBJECT (demux, "combined flow return: %s", gst_flow_get_name (ret));
   return ret;
 }
 
@@ -1350,8 +1354,9 @@ gst_qtdemux_loop_state_movie (GstQTDemux * qtdemux)
 
     GST_LOG_OBJECT (qtdemux,
         "Pushing buffer with time %" GST_TIME_FORMAT ", duration %"
-        GST_TIME_FORMAT " on pad %p", GST_TIME_ARGS (timestamp),
-        GST_TIME_ARGS (duration), stream->pad);
+        GST_TIME_FORMAT " on pad %s",
+        GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
+        GST_PAD_NAME (stream->pad));
     ret = gst_pad_push (stream->pad, buf);
   } else {
     ret = GST_FLOW_OK;
@@ -1464,7 +1469,7 @@ pause:
 
 /*
  * next_entry_size
- * 
+ *
  * Returns the size of the first entry at the current offset.
  * If -1, there are none (which means EOS or empty file).
  */
@@ -2613,7 +2618,7 @@ done:
 
 /* parse the traks.
  * With each track we associate a new QtDemuxStream that contains all the info
- * about the trak. 
+ * about the trak.
  * traks that do not decode to something (like strm traks) will not have a pad.
  */
 static gboolean
@@ -2667,7 +2672,7 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
     stream->duration = QT_UINT32 ((guint8 *) mdhd->data + 24);
   }
 
-  GST_LOG_OBJECT (qtdemux, "track timescale: %" G_GUINT64_FORMAT,
+  GST_LOG_OBJECT (qtdemux, "track timescale: %" G_GUINT32_FORMAT,
       stream->timescale);
   GST_LOG_OBJECT (qtdemux, "track duration: %" G_GUINT64_FORMAT,
       stream->duration);
@@ -2685,7 +2690,8 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
      * identify those yet, except for just looking at their duration. */
     if (tdur1 != 0 && (tdur2 * 10 / tdur1) < 2) {
       GST_WARNING_OBJECT (qtdemux,
-          "Track shorter than 20%% (%d/%d vs. %d/%d) of the stream "
+          "Track shorter than 20%% (%" G_GUINT64_FORMAT "/%" G_GUINT32_FORMAT
+          " vs. %" G_GUINT32_FORMAT "/%" G_GUINT32_FORMAT ") of the stream "
           "found, assuming preview image or something; skipping track",
           stream->duration, stream->timescale, qtdemux->duration,
           qtdemux->timescale);
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 9f5d58d8..74907912 100644
--- 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 7af74671..8007d54b 100644
--- 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];
diff --git a/gst/switch/Makefile.am b/gst/switch/Makefile.am
index fcac882e..b5a55038 100644
--- a/gst/switch/Makefile.am
+++ b/gst/switch/Makefile.am
@@ -4,6 +4,6 @@ plugin_LTLIBRARIES = libgstswitch.la
 libgstswitch_la_SOURCES = gstswitch.c
 libgstswitch_la_CFLAGS = $(GST_CFLAGS)
 libgstswitch_la_LIBADD =
-libgstswitch_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstswitch_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS)
 
 noinst_HEADERS = gstswitch.h