audiosink/RenderAudioSamples: sync w/ pipe clock, drop late

diff --git a/docs/plugins/scanobj-build.stamp b/docs/plugins/scanobj-build.stamp
deleted file mode 100644
index e69de29..0000000
diff --git a/sys/decklink/gstdecklinksink.cpp b/sys/decklink/gstdecklinksink.cpp
index 257277d..657cf4e 100644
--- a/sys/decklink/gstdecklinksink.cpp
+++ b/sys/decklink/gstdecklinksink.cpp
@@ -696,24 +696,76 @@ gst_decklink_sink_audiosink_chain (GstPad * pad, GstObject * parent,
 {
   GstDecklinkSink *decklinksink;
   GstFlowReturn ret;
-
+  GstClockTime timestamp, duration;
+  GstClockTime running_time;
+  GstClockTime running_time_duration;
+  GstClock *clock;
+  GstClockID clock_id;
+  GstClockTime base_time;
   decklinksink = GST_DECKLINK_SINK (parent);

   if (decklinksink->stop)
     return GST_FLOW_FLUSHING;

-  g_mutex_lock (&decklinksink->audio_mutex);
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  duration = GST_BUFFER_DURATION (buffer);
+  if (duration == GST_CLOCK_TIME_NONE) {
+    duration = gst_util_uint64_scale_int (GST_SECOND, gst_buffer_get_size (buffer), 2 * 2 *48000);
+    GST_WARNING_OBJECT (decklinksink, "no audio duration, setting it to %lu",
+        duration);
+  }
+
+  running_time =
+      gst_segment_to_running_time (&decklinksink->audio_segment,
+      GST_FORMAT_TIME, timestamp);
+  running_time_duration =
+      gst_segment_to_running_time (&decklinksink->audio_segment,
+      GST_FORMAT_TIME, timestamp + duration) - running_time;
+
+  clock = gst_element_get_clock (GST_ELEMENT (decklinksink));
+  base_time = gst_element_get_base_time (GST_ELEMENT (decklinksink));
+  if (clock) {
+    if (base_time != GST_CLOCK_TIME_NONE) {
+      GstClockTime clock_time = gst_clock_get_time (clock);
+
+      if (base_time + running_time + running_time_duration <
+          gst_clock_get_time (clock)) {
+        GST_WARNING_OBJECT (decklinksink,
+            "Dropping too late frame: %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (running_time + running_time_duration),
+            GST_TIME_ARGS (clock_time - base_time));
+        gst_object_unref (clock);
+        //flow_ret = GST_FLOW_OK;
+        goto out;
+      }
+
+      if (base_time + running_time > gst_clock_get_time(clock) ) {
+        clock_id = gst_clock_new_single_shot_id (clock, base_time + running_time);
+        GST_WARNING_OBJECT (decklinksink,
+            "waiting for clock: %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (clock_time), GST_TIME_ARGS (base_time + running_time));
+        gst_clock_id_wait (clock_id, NULL);
+        //GST_LOG_OBJECT (decklinksink, "finished waiting for clock");
+        gst_clock_id_unref (clock_id);
+      }
+    }
+
+    gst_object_unref (clock);
+  }
+
   /* FIXME: This is not correct as it assumes that the decklink clock
    * has the exact same value as the pipeline clock at any time. Not
    * only the same rate!
    */
-  while (!decklinksink->stop &&
-      gst_adapter_available (decklinksink->audio_adapter) > 1600 * 4 * 2) {
-    g_cond_wait (&decklinksink->audio_cond, &decklinksink->audio_mutex);
-  }
+  //while (!decklinksink->stop &&
+    //  gst_adapter_available (decklinksink->audio_adapter) > 1600 * 4 * 2) {
+    //g_cond_wait (&decklinksink->audio_cond, &decklinksink->audio_mutex);
+  //}
+  g_mutex_lock (&decklinksink->audio_mutex);
   gst_adapter_push (decklinksink->audio_adapter, buffer);
   g_mutex_unlock (&decklinksink->audio_mutex);

+out:
   ret = GST_FLOW_OK;
   return ret;
 }
@@ -795,6 +847,13 @@ Output::RenderAudioSamples (bool preroll)
 {
   uint32_t samplesWritten;
   HRESULT ret;
+  GstClockTime timestamp, duration;
+  GstClockTime running_time;
+  GstClockTime running_time_duration;
+  GstClock *clock;
+  GstClockID clock_id;
+  GstClockTime base_time;
+  guint64 dist;

   // guint64 samplesToWrite;

@@ -806,18 +865,67 @@ Output::RenderAudioSamples (bool preroll)
     gconstpointer data;
     int n;

+
     g_mutex_lock (&decklinksink->audio_mutex);

     n = gst_adapter_available (decklinksink->audio_adapter);
     if (n > 0) {
       data = gst_adapter_map (decklinksink->audio_adapter, n);

+      clock = gst_element_get_clock (GST_ELEMENT (decklinksink));
+      base_time = gst_element_get_base_time (GST_ELEMENT (decklinksink));
+      timestamp = gst_adapter_prev_pts (decklinksink->audio_adapter, &dist);
+
+      if (timestamp == GST_CLOCK_TIME_NONE) {
+        GST_WARNING_OBJECT (decklinksink,
+            "No timestamp for audio adapter");
+      }
+
+      running_time =
+        gst_segment_to_running_time (&decklinksink->audio_segment,
+            GST_FORMAT_TIME, timestamp);
+      duration = gst_util_uint64_scale_int (GST_SECOND, 48000, 1);
+
+      running_time_duration =
+        gst_segment_to_running_time (&decklinksink->audio_segment,
+            GST_FORMAT_TIME, timestamp + duration) - running_time;
+
+      if (clock) {
+        if (base_time != GST_CLOCK_TIME_NONE) {
+          GstClockTime clock_time = gst_clock_get_time (clock);
+
+          if (base_time + running_time + running_time_duration <
+              gst_clock_get_time (clock)) {
+            GST_WARNING_OBJECT (decklinksink,
+                "Dropping too late frame (avail=%d, dist=%lu dropping=%ld) : %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
+                n, dist, (gint64)n - (gint64)dist,
+                GST_TIME_ARGS (running_time + running_time_duration),
+                GST_TIME_ARGS (clock_time - base_time));
+            gst_object_unref (clock);
+            gst_adapter_flush (decklinksink->audio_adapter, n - dist);
+            return S_OK;
+          }
+
+          if (base_time + running_time > gst_clock_get_time(clock) ) {
+            clock_id = gst_clock_new_single_shot_id (clock, base_time + running_time);
+            GST_LOG_OBJECT (decklinksink,
+                "waiting for clock: %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT,
+                GST_TIME_ARGS (clock_time), GST_TIME_ARGS (base_time + running_time));
+            gst_clock_id_wait (clock_id, NULL);
+            GST_LOG_OBJECT (decklinksink, "finished waiting for clock");
+            gst_clock_id_unref (clock_id);
+          }
+        }
+
+        gst_object_unref (clock);
+      }
+
       /* FIXME: This is not correct. We assume here that the decklink
        * clock runs at the same rate as the pipeline clock and that
        * the input stream has no discontinuities at all
        */
       ret = decklinksink->output->ScheduleAudioSamples ((void *) data, n / 4,
-          0, 0, &samplesWritten);
+          running_time, GST_SECOND, &samplesWritten);

       gst_adapter_unmap (decklinksink->audio_adapter);
       gst_adapter_flush (decklinksink->audio_adapter, samplesWritten * 4);