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

diff --git a/gst/audiomixer/gstaudioaggregator.c b/gst/audiomixer/gstaudioaggregator.c
index 01704f5..38e8709 100644 (file)
--- a/gst/audiomixer/gstaudioaggregator.c
+++ b/gst/audiomixer/gstaudioaggregator.c
@@ -739,7 +739,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
   GstClockTime start_time, end_time;
   gboolean discont = FALSE;
   guint64 start_offset, end_offset;
-  GstClockTime timestamp, stream_time = GST_CLOCK_TIME_NONE;
   gint rate, bpf;
 
   GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
@@ -762,15 +761,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
     goto done;
   }
 
-  timestamp = GST_BUFFER_PTS (inbuf);
-  stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
-      timestamp);
-
-  /* sync object properties on stream time */
-  /* TODO: Ideally we would want to do that on every sample */
-  if (GST_CLOCK_TIME_IS_VALID (stream_time))
-    gst_object_sync_values (GST_OBJECT (pad), stream_time);
-
   start_time = GST_BUFFER_PTS (inbuf);
   end_time =
       start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
@@ -964,6 +954,29 @@ gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
   return outbuf;
 }
 
+static gboolean
+sync_pad_values (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad)
+{
+  GstAggregatorPad *bpad = GST_AGGREGATOR_PAD (pad);
+  GstClockTime timestamp, stream_time;
+
+  if (pad->priv->buffer == NULL)
+    return TRUE;
+
+  timestamp = GST_BUFFER_PTS (pad->priv->buffer);
+  GST_OBJECT_LOCK (bpad);
+  stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
+      timestamp);
+  GST_OBJECT_UNLOCK (bpad);
+
+  /* sync object properties on stream time */
+  /* TODO: Ideally we would want to do that on every sample */
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (GST_OBJECT (pad), stream_time);
+
+  return TRUE;
+}
+
 static GstFlowReturn
 gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
 {
@@ -1011,6 +1024,10 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
   element = GST_ELEMENT (agg);
   aagg = GST_AUDIO_AGGREGATOR (agg);
 
+  /* Sync pad properties to the stream time */
+  gst_aggregator_iterate_sinkpads (agg,
+      (GstAggregatorPadForeachFunc) sync_pad_values, NULL);
+
   GST_AUDIO_AGGREGATOR_LOCK (aagg);
   GST_OBJECT_LOCK (agg);
 

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 233302c..7aef48e 100644 (file)
--- 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 ddfdbe1..8fe329d 100644 (file)
--- 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