decklinkaudiosrc: Extrapolate stream/hw reference timestamps when video frame is missing

Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.

https://bugzilla.gnome.org/show_bug.cgi?id=792042

2 files changed, 25 insertions, 2 deletions

diff --git a/sys/decklink/gstdecklinkaudiosrc.cpp b/sys/decklink/gstdecklinkaudiosrc.cpp
index 10fb1ed20..887ad0ff2 100644
--- a/sys/decklink/gstdecklinkaudiosrc.cpp
+++ b/sys/decklink/gstdecklinkaudiosrc.cpp
@@ -707,6 +707,27 @@ retry:
 
   // Detect gaps in stream time
   self->processed += sample_count;
+  if (self->expected_stream_time != GST_CLOCK_TIME_NONE
+      && p.stream_timestamp == GST_CLOCK_TIME_NONE && !p.no_signal) {
+    /* We missed a frame. Extrapolate the timestamps */
+    p.stream_timestamp = self->expected_stream_time;
+    p.stream_duration =
+        gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
+  }
+  if (self->last_hardware_time != GST_CLOCK_TIME_NONE
+      && p.hardware_timestamp == GST_CLOCK_TIME_NONE && !p.no_signal) {
+    /* This should always happen when the previous one also does, but let's
+     * have two separate checks just in case */
+    GstClockTime start_hw_offset, end_hw_offset;
+    start_hw_offset =
+        gst_util_uint64_scale (self->last_hardware_time, self->info.rate,
+        GST_SECOND);
+    end_hw_offset = start_hw_offset + sample_count;
+    p.hardware_timestamp =
+        gst_util_uint64_scale_int (end_hw_offset, GST_SECOND, self->info.rate);
+    /* Will be the same as the stream duration - reuse it */
+    p.hardware_duration = p.stream_duration;
+  }
 
   if (p.stream_timestamp != GST_CLOCK_TIME_NONE) {
     GstClockTime start_stream_time, end_stream_time;
@@ -734,14 +755,15 @@ retry:
           GST_FORMAT_TIME, timestamp);
 
       msg =
-          gst_message_new_qos (GST_OBJECT (self), TRUE, running_time, p.stream_timestamp,
-          timestamp, duration);
+          gst_message_new_qos (GST_OBJECT (self), TRUE, running_time,
+          p.stream_timestamp, timestamp, duration);
       gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, self->processed,
           self->dropped);
       gst_element_post_message (GST_ELEMENT (self), msg);
     }
     self->expected_stream_time = end_stream_time;
   }
+  self->last_hardware_time = p.hardware_timestamp;
 
   if (p.no_signal)
     GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_GAP);
diff --git a/sys/decklink/gstdecklinkaudiosrc.h b/sys/decklink/gstdecklinkaudiosrc.h
index c631d39e6..405da7ae5 100644
--- a/sys/decklink/gstdecklinkaudiosrc.h
+++ b/sys/decklink/gstdecklinkaudiosrc.h
@@ -75,6 +75,7 @@ struct _GstDecklinkAudioSrc
   GstClockTime expected_stream_time;
   guint64 processed;
   guint64 dropped;
+  GstClockTime last_hardware_time;
 
   /* Last time we noticed a discont */
   GstClockTime discont_time;