Merge branch 'master' into 0.11

Conflicts:
	libs/gst/base/gstbaseparse.c
	libs/gst/base/gstbasesink.c

docs/manual/advanced-clocks.xml

@@ -12,14 +12,16 @@
   </para>
   
   <para>
-    &GStreamer; derives several times from the clock and the playback state.
+    &GStreamer; derives several <emphasis>time value</emphasis> from the clock
+    and the playback state.
     It is important to note, that a  <emphasis>clock-time</emphasis> is
     monotonically rising, but the value itself is not meaningful.
     Subtracting the <emphasis>base-time</emphasis> yields the
     <emphasis>running-time</emphasis>. It is the same as the
     <emphasis>stream-time</emphasis> if one plays from start to end at original
     rate. The <emphasis>stream-time</emphasis> indicates the position in the
-    media.
+    media. The <emphasis>running-time</emphasis> is (re-)set to 0 when the 
+    pipeline starts to play and also after <emphasis>flushing</emphasis> seeks.
   </para>
 
   <figure float="1" id="chapter-clock-img">

gst/gstevent.c

@@ -409,7 +409,7 @@ gst_event_has_name (GstEvent * event, const gchar * name)
  * required.
  *
  * Note that events and messages share the same sequence number incrementor;
- * two events or messages will never not have the same sequence number unless
+ * two events or messages will never have the same sequence number unless
  * that correspondence was made explicitly.
  *
  * Returns: The event's sequence number.

gst/gstmessage.c

@@ -309,7 +309,7 @@ gst_message_new_custom (GstMessageType type, GstObject * src,
  * it is not required.
  *
  * Note that events and messages share the same sequence number incrementor;
- * two events or messages will never not have the same sequence number unless
+ * two events or messages will never have the same sequence number unless
  * that correspondence was made explicitly.
  *
  * Returns: The message's sequence number.

libs/gst/base/gstbaseparse.c

@@ -1935,9 +1935,10 @@ gst_base_parse_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
     ret = GST_FLOW_OK;
   } else if (ret == GST_FLOW_OK) {
     if (parse->segment.rate > 0.0) {
+      GST_LOG_OBJECT (parse, "pushing frame (%" G_GSIZE_FORMAT " bytes) now..",
+          size);
       ret = gst_pad_push (parse->srcpad, buffer);
-      GST_LOG_OBJECT (parse, "frame (%" G_GSIZE_FORMAT " bytes) pushed: %s",
-          size, gst_flow_get_name (ret));
+      GST_LOG_OBJECT (parse, "frame pushed, flow %s", gst_flow_get_name (ret));
     } else {
       GST_LOG_OBJECT (parse, "frame (%" G_GSIZE_FORMAT " bytes) queued for now",
           size);
@@ -1946,9 +1947,9 @@ gst_base_parse_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
       ret = GST_FLOW_OK;
     }
   } else {
-    gst_buffer_unref (buffer);
     GST_LOG_OBJECT (parse, "frame (%" G_GSIZE_FORMAT " bytes) not pushed: %s",
         size, gst_flow_get_name (ret));
+    gst_buffer_unref (buffer);
     /* if we are not sufficiently in control, let upstream decide on EOS */
     if (ret == GST_FLOW_UNEXPECTED &&
         (parse->priv->passthrough ||

libs/gst/base/gstbasesink.c

@@ -227,6 +227,8 @@ struct _GstBaseSinkPrivate
 
   /* if we already commited the state */
   gboolean commited;
+  /* state change to playing ongoing */
+  gboolean to_playing;
 
   /* when we received EOS */
   gboolean received_eos;
@@ -4503,6 +4505,14 @@ gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format,
   else
     gst_object_ref (clock);
 
+  /* mainloop might be querying position when going to playing async,
+   * while (audio) rendering might be quickly advancing stream position,
+   * so use clock asap rather than last reported position */
+  if (in_paused && with_clock && g_atomic_int_get (&basesink->priv->to_playing)) {
+    GST_DEBUG_OBJECT (basesink, "going to PLAYING, so not PAUSED");
+    in_paused = FALSE;
+  }
+
   /* collect all data we need holding the lock */
   if (GST_CLOCK_TIME_IS_VALID (segment->time))
     time = segment->time;
@@ -4907,6 +4917,7 @@ gst_base_sink_change_state (GstElement * element, GstStateChange transition)
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       GST_BASE_SINK_PREROLL_LOCK (basesink);
+      g_atomic_int_set (&basesink->priv->to_playing, TRUE);
       if (!gst_base_sink_needs_preroll (basesink)) {
         GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll");
         /* no preroll needed anymore now. */
@@ -4952,7 +4963,14 @@ gst_base_sink_change_state (GstElement * element, GstStateChange transition)
   }
 
   switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      /* completed transition, so need not be marked any longer
+       * And it should be unmarked, since e.g. losing our position upon flush
+       * does not really change state to PAUSED ... */
+      g_atomic_int_set (&basesink->priv->to_playing, FALSE);
+      break;
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      g_atomic_int_set (&basesink->priv->to_playing, FALSE);
       GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED");
       /* FIXME, make sure we cannot enter _render first */
 

tests/check/elements/filesrc.c

@@ -443,6 +443,9 @@ GST_START_TEST (test_uri_query)
     check_uri_for_location (src, "./foo", NULL);
     check_uri_for_location (src, "../foo", NULL);
     check_uri_for_location (src, "foo/./bar", NULL);
+    /* make sure non-ASCII characters are escaped properly (U+00F6 here) */
+    check_uri_for_location (src, "/i/./d\303\266/not/../exist",
+        "file:///i/d%C3%B6/exist");
   }
 #endif