gst/playback/gstplaybasebin.c: Small debug improvement.

Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.

diff --git a/ChangeLog b/ChangeLog
index d5940ab..06e9ec3 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,16 @@
 2007-06-28  Wim Taymans  <wim@fluendo.com>
 
+       * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
+       Small debug improvement.
+
+       * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
+       (plugin_init):
+       Tweak the rate estimation period.
+       When calculating the buffer filledness in rate estimation mode, don't
+       mix it with other metrics.
+
+2007-06-28  Wim Taymans  <wim@fluendo.com>
+
        * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
        (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
        When creating the groups, allow for a 5 second, unlimited buffers

diff --git a/gst/playback/gstplaybasebin.c b/gst/playback/gstplaybasebin.c
index bcb7fc8..0a8998c 100644 (file)
--- a/gst/playback/gstplaybasebin.c
+++ b/gst/playback/gstplaybasebin.c
@@ -1899,6 +1899,9 @@ make_decoder (GstPlayBaseBin * play_base_bin)
   g_object_set_data (G_OBJECT (decoder), "pending", "1");
   play_base_bin->pending++;
 
+  GST_DEBUG_OBJECT (play_base_bin, "created decodebin, %d pending",
+      play_base_bin->pending);
+
   play_base_bin->decoders = g_slist_prepend (play_base_bin->decoders, decoder);
 
   return decoder;
@@ -2080,7 +2083,6 @@ setup_source (GstPlayBaseBin * play_base_bin)
     return TRUE;
   }
   if (is_dynamic) {
-    GST_DEBUG_OBJECT (play_base_bin, "Source has dynamic output pads");
     /* connect a handler for the new-pad signal */
     play_base_bin->src_np_sig_id =
         g_signal_connect (G_OBJECT (play_base_bin->source), "pad-added",
@@ -2090,6 +2092,8 @@ setup_source (GstPlayBaseBin * play_base_bin)
         G_CALLBACK (source_no_more_pads), play_base_bin);
     g_object_set_data (G_OBJECT (play_base_bin->source), "pending", "1");
     play_base_bin->pending++;
+    GST_DEBUG_OBJECT (play_base_bin,
+        "Source has dynamic output pads, %d pending", play_base_bin->pending);
   } else {
     GstElement *decoder;
 

diff --git a/gst/playback/gstqueue2.c b/gst/playback/gstqueue2.c
index 285decc..1bbe495 100644 (file)
--- a/gst/playback/gstqueue2.c
+++ b/gst/playback/gstqueue2.c
@@ -588,6 +588,16 @@ apply_segment (GstQueue * queue, GstEvent * event, GstSegment * segment)
   gst_event_parse_new_segment_full (event, &update, &rate, &arate,
       &format, &start, &stop, &time);
 
+  GST_DEBUG_OBJECT (queue,
+      "received NEWSEGMENT update %d, rate %lf, applied rate %lf, "
+      "format %d, "
+      "%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %"
+      G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate,
+      format, start, stop, time);
+
+  if (format == GST_FORMAT_BYTES) {
+  }
+
   /* now configure the values, we use these to track timestamps on the
    * sinkpad. */
   if (format != GST_FORMAT_TIME) {
@@ -654,11 +664,13 @@ update_buffering (GstQueue * queue)
     percent = 100;
   } else {
     /* figure out the percent we are filled, we take the max of all formats. */
-    percent = GET_PERCENT (bytes);
-    percent = MAX (percent, GET_PERCENT (time));
-    percent = MAX (percent, GET_PERCENT (buffers));
-    if (queue->use_rate_estimate)
-      percent = MAX (percent, GET_PERCENT (rate_time));
+    if (queue->use_rate_estimate) {
+      percent = GET_PERCENT (rate_time);
+    } else {
+      percent = GET_PERCENT (bytes);
+      percent = MAX (percent, GET_PERCENT (time));
+      percent = MAX (percent, GET_PERCENT (buffers));
+    }
   }
 
   if (queue->is_buffering) {
@@ -702,11 +714,12 @@ reset_rate_timer (GstQueue * queue)
   queue->timer_started = FALSE;
 }
 
+/* the interval in seconds to recalculate the rate */
+#define RATE_INTERVAL    0.2
 /* Tuning for rate estimation. We use a large window for the input rate because
  * it should be stable when connected to a network. The output rate is less
  * stable (the elements preroll, queues behind a demuxer fill, ...) and should
  * therefore adapt more quickly. */
-#define RATE_INTERVAL    0.5
 #define AVG_IN(avg,val)  ((avg) * 15.0 + (val)) / 16.0
 #define AVG_OUT(avg,val) ((avg) * 3.0 + (val)) / 4.0
 
@@ -1825,8 +1838,8 @@ gst_queue_get_property (GObject * object,
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  GST_DEBUG_CATEGORY_INIT (queue_debug, "queue", 0, "queue element");
-  GST_DEBUG_CATEGORY_INIT (queue_dataflow, "queue_dataflow", 0,
+  GST_DEBUG_CATEGORY_INIT (queue_debug, "queue2", 0, "queue element");
+  GST_DEBUG_CATEGORY_INIT (queue_dataflow, "queue2_dataflow", 0,
       "dataflow inside the queue element");
 
 #ifdef ENABLE_NLS