audioaggregator: Sync pad values before aggregating

We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.

Also add a test for this.

https://bugzilla.gnome.org/show_bug.cgi?id=749574

2 files changed, 78 insertions, 2 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 233302ce1..7aef48ea0 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -296,8 +296,8 @@ AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS) \
 	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
 LDADD = $(GST_CHECK_LIBS)
 
-elements_audiomixer_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
-elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
+elements_audiomixer_LDADD = $(GST_BASE_LIBS) $(GST_CONTROLLER_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
+elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CONTROLLER_CFLAGS) $(AM_CFLAGS)
 
 elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ -lgstaudio-@GST_API_VERSION@ $(LDADD)
 elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c
index ddfdbe188..8fe329d82 100644
--- a/tests/check/elements/audiomixer.c
+++ b/tests/check/elements/audiomixer.c
@@ -35,6 +35,8 @@
 #include <gst/check/gstconsistencychecker.h>
 #include <gst/audio/audio.h>
 #include <gst/base/gstbasesrc.h>
+#include <gst/controller/gstdirectcontrolbinding.h>
+#include <gst/controller/gstinterpolationcontrolsource.h>
 
 static GMainLoop *main_loop;
 
@@ -1835,6 +1837,79 @@ GST_START_TEST (test_segment_base_handling)
 
 GST_END_TEST;
 
+static void
+set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
+    GstClockTime end, gdouble end_value)
+{
+  GstControlSource *cs;
+  GstTimedValueControlSource *tvcs;
+
+  cs = gst_interpolation_control_source_new ();
+  fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
+          gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
+              "volume", cs)));
+
+  /* set volume interpolation mode */
+  g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
+
+  tvcs = (GstTimedValueControlSource *) cs;
+  fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
+  fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
+  gst_object_unref (cs);
+}
+
+GST_START_TEST (test_sinkpad_property_controller)
+{
+  GstBus *bus;
+  GstMessage *msg;
+  GstElement *pipeline, *sink, *mix, *src1;
+  GstPad *srcpad, *sinkpad;
+  GError *error = NULL;
+  gchar *debug;
+
+  pipeline = gst_pipeline_new ("pipeline");
+  mix = gst_element_factory_make ("audiomixer", "audiomixer");
+  sink = gst_element_factory_make ("fakesink", "sink");
+  src1 = gst_element_factory_make ("audiotestsrc", "src1");
+  g_object_set (src1, "num-buffers", 100, NULL);
+  gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
+  fail_unless (gst_element_link (mix, sink));
+
+  srcpad = gst_element_get_static_pad (src1, "src");
+  sinkpad = gst_element_get_request_pad (mix, "sink_0");
+  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
+  set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
+  gst_object_unref (sinkpad);
+  gst_object_unref (srcpad);
+
+  gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+  msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
+      GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_ERROR:
+      gst_message_parse_error (msg, &error, &debug);
+      g_printerr ("ERROR from element %s: %s\n",
+          GST_OBJECT_NAME (msg->src), error->message);
+      g_printerr ("Debug info: %s\n", debug);
+      g_error_free (error);
+      g_free (debug);
+      break;
+    case GST_MESSAGE_EOS:
+      break;
+    default:
+      g_assert_not_reached ();
+  }
+  gst_message_unref (msg);
+  g_object_unref (bus);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
 static Suite *
 audiomixer_suite (void)
 {
@@ -1859,6 +1934,7 @@ audiomixer_suite (void)
   tcase_add_test (tc_chain, test_sync_discont);
   tcase_add_test (tc_chain, test_sync_unaligned);
   tcase_add_test (tc_chain, test_segment_base_handling);
+  tcase_add_test (tc_chain, test_sinkpad_property_controller);
 
   /* Use a longer timeout */
 #ifdef HAVE_VALGRIND