Merge branch 'master' into 0.11

common

@@ -1 +1 @@
-Subproject commit 1de7f6ab2d4bc1af69f06079cf0f4e2cbbfdc178
+Subproject commit 6aec6b9716c184c60c4bc6a5916a2471cfa8c8cd

configure.ac

@@ -906,7 +906,7 @@ AC_SUBST(GST_LIB_LDFLAGS)
 
 dnl this really should only contain flags, not libs - they get added before
 dnl whatevertarget_LIBS and -L flags here affect the rest of the linking
-GST_PLUGIN_LDFLAGS="-module -avoid-version -export-symbols-regex '^[_]*gst_plugin_desc\$\$' $GST_ALL_LDFLAGS"
+GST_PLUGIN_LDFLAGS="-module -avoid-version -export-symbols-regex '^[_]*gst_plugin_desc.*' $GST_ALL_LDFLAGS"
 AC_SUBST(GST_PLUGIN_LDFLAGS)
 
 dnl *** output files ***

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -56,7 +56,6 @@ struct _GstBaseAudioSinkPrivate
   gboolean sync_latency;
 
   GstClockTime eos_time;
-  gint eos_rendering;
 
   gboolean do_time_offset;
   /* number of microseconds we alow timestamps or clock slaving to drift
@@ -841,10 +840,6 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
   if (!sink->ringbuffer->spec.rate)
     return TRUE;
 
-  /* if PLAYING is interrupted,
-   * arrange to have clock running when going to PLAYING again */
-  g_atomic_int_set (&sink->priv->eos_rendering, 1);
-
   /* need to start playback before we can drain, but only when
    * we have successfully negotiated a format and thus acquired the
    * ringbuffer. */
@@ -862,7 +857,6 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
 
     GST_DEBUG_OBJECT (sink, "drained audio");
   }
-  g_atomic_int_set (&sink->priv->eos_rendering, 0);
   return TRUE;
 }
 
@@ -1894,7 +1888,6 @@ gst_base_audio_sink_change_state (GstElement * element,
       sink->next_sample = -1;
       sink->priv->last_align = -1;
       sink->priv->eos_time = -1;
-      sink->priv->eos_rendering = 0;
       gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
       gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
 
@@ -1909,19 +1902,24 @@ gst_base_audio_sink_change_state (GstElement * element,
                 sink->provided_clock, TRUE));
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+    {
+      gboolean eos;
+
       GST_OBJECT_LOCK (sink);
       GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
       sink->priv->sync_latency = TRUE;
+      eos = GST_BASE_SINK (sink)->eos;
       GST_OBJECT_UNLOCK (sink);
 
       gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
-      if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL ||
-          g_atomic_int_get (&sink->priv->eos_rendering)) {
+      if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL || eos) {
         /* we always start the ringbuffer in pull mode immediatly */
-        /* sync rendering on eos needs running clock */
+        /* sync rendering on eos needs running clock,
+         * and others need running clock when finished rendering eos */
         gst_ring_buffer_start (sink->ringbuffer);
       }
       break;
+    }
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
       /* ringbuffer cannot start anymore */
       gst_ring_buffer_may_start (sink->ringbuffer, FALSE);

gst/playback/gstplaybin2.c

@@ -3194,31 +3194,21 @@ activate_group (GstPlayBin * playbin, GstSourceGroup * group, GstState target)
     group->uridecodebin = gst_object_ref (uridecodebin);
   }
 
-  /* configure connection speed */
-  g_object_set (uridecodebin, "connection-speed",
-      playbin->connection_speed / 1000, NULL);
-
   flags = gst_play_sink_get_flags (playbin->playsink);
 
-  /* configure download buffering */
-  if (flags & GST_PLAY_FLAG_DOWNLOAD)
-    g_object_set (uridecodebin, "download", TRUE, NULL);
-  else
-    g_object_set (uridecodebin, "download", FALSE, NULL);
-
-  /* configure uri */
-  g_object_set (uridecodebin, "uri", group->uri, NULL);
-  /* configure buffering of demuxed/parsed data */
-  if (flags & GST_PLAY_FLAG_BUFFERING)
-    g_object_set (uridecodebin, "use-buffering", TRUE, NULL);
-  else
-    g_object_set (uridecodebin, "use-buffering", FALSE, NULL);
-  /* configure buffering parameters */
-  g_object_set (uridecodebin, "buffer-duration", playbin->buffer_duration,
-      NULL);
-  g_object_set (uridecodebin, "buffer-size", playbin->buffer_size, NULL);
-  g_object_set (uridecodebin, "ring-buffer-max-size",
-      playbin->ring_buffer_max_size, NULL);
+  g_object_set (uridecodebin,
+      /* configure connection speed */
+      "connection-speed", playbin->connection_speed / 1000,
+      /* configure uri */
+      "uri", group->uri,
+      /* configure download buffering */
+      "download", ((flags & GST_PLAY_FLAG_DOWNLOAD) != 0),
+      /* configure buffering of demuxed/parsed data */
+      "use-buffering", ((flags & GST_PLAY_FLAG_BUFFERING) != 0),
+      /* configure buffering parameters */
+      "buffer-duration", playbin->buffer_duration,
+      "buffer-size", playbin->buffer_size,
+      "ring-buffer-max-size", playbin->ring_buffer_max_size, NULL);
 
   /* connect pads and other things */
   group->pad_added_id = g_signal_connect (uridecodebin, "pad-added",
@@ -3268,11 +3258,11 @@ activate_group (GstPlayBin * playbin, GstSourceGroup * group, GstState target)
       group->suburidecodebin = gst_object_ref (suburidecodebin);
     }
 
-    /* configure connection speed */
-    g_object_set (suburidecodebin, "connection-speed",
-        playbin->connection_speed, NULL);
-    /* configure uri */
-    g_object_set (suburidecodebin, "uri", group->suburi, NULL);
+    g_object_set (suburidecodebin,
+        /* configure connection speed */
+        "connection-speed", playbin->connection_speed,
+        /* configure uri */
+        "uri", group->suburi, NULL);
 
     /* connect pads and other things */
     group->sub_pad_added_id = g_signal_connect (suburidecodebin, "pad-added",

gst/typefind/gsttypefindfunctions.c

@@ -2149,7 +2149,7 @@ h263_video_type_find (GstTypeFind * tf, gpointer unused)
 
     /* Find the picture start code */
     data = (data << 8) + c.data[0];
-    psc = data & 0xfffffc0000;
+    psc = data & G_GUINT64_CONSTANT (0xfffffc0000);
     if (psc == 0x800000) {
       /* Found PSC */
       /* TR */