audiotestsrc: Fix the way we compute EOS in reverse playback

In reverse playback we were not taking into account the current buffer
samples to check if we had reached EOS which was leading to a buffer
with PTS = CLOCK_TIME_NONE containing too many frames followed by a
useless buffer with pts=0 duration=0, and a g_critical issue in
gst_object_sync_values.

Also add a validate based test case.
Without that patch this is how the expectation fails:

``` diff
--- log-asink-sink-expected       2020-05-22 23:22:42.654384579 -0400
+++ log-asink-sink-actual  2020-05-22 23:29:35.671586380 -0400
@@ -27,5 +27,6 @@
 buffer: pts=0:00:00.058820861, due=0:00:00.023219955, flags=discont
 buffer: pts=0:00:00.035600907, due=0:00:00.023219954, flags=discont
 buffer: pts=0:00:00.012380952, due=0:00:00.023219955, flags=discont
-buffer: pts=0:00:00.000000000, due=0:00:00.012380952, flags=discont
+buffer: due=0:00:00.012380953, flags=discont
+buffer: pts=0:00:00.000000000, flags=discont
 event eos: (no structure)
 ```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/667>

1 files changed, 2 insertions, 2 deletions

diff --git a/gst/audiotestsrc/gstaudiotestsrc.c b/gst/audiotestsrc/gstaudiotestsrc.c
index b06b336cc..00645eb20 100644
--- a/gst/audiotestsrc/gstaudiotestsrc.c
+++ b/gst/audiotestsrc/gstaudiotestsrc.c
@@ -1489,10 +1489,10 @@ gst_audio_test_src_fill (GstBaseSrc * basesrc, guint64 offset,
     next_sample = src->sample_stop;
     src->eos_reached = TRUE;
   } else if (src->check_seek_stop && src->reverse &&
-      (src->sample_stop > src->next_sample)
+      (src->sample_stop >= (src->next_sample - samples))
       ) {
     /* calculate only partial buffer */
-    src->generate_samples_per_buffer = src->sample_stop - src->next_sample;
+    src->generate_samples_per_buffer = src->next_sample - src->sample_stop;
     next_sample = src->sample_stop;
     src->eos_reached = TRUE;
   } else {