audioresample: Add new test that checks for downstream renegotiation

This test always consumes 48kHz and outputs different sample rates based
on downstream renegotiation. Previously this would produce completely
wrong timestamps and not output all samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>

diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/audioresample.c
index 77dda39..793594b 100644 (file)
--- a/tests/check/elements/audioresample.c
+++ b/tests/check/elements/audioresample.c
@@ -570,6 +570,179 @@ GST_START_TEST (test_live_switch)
 
 GST_END_TEST;
 
+static gint current_rate = 0;
+
+static gboolean
+live_switch_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
+{
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_ACCEPT_CAPS:{
+      GstCaps *acceptable_caps;
+      GstCaps *caps;
+
+      acceptable_caps = gst_pad_get_current_caps (mysrcpad);
+      acceptable_caps = gst_caps_make_writable (acceptable_caps);
+      gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
+          NULL);
+
+      gst_query_parse_accept_caps (query, &caps);
+
+      gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps,
+              acceptable_caps));
+
+      gst_caps_unref (acceptable_caps);
+
+      return TRUE;
+    }
+    case GST_QUERY_CAPS:{
+      GstCaps *acceptable_caps;
+      GstCaps *filter;
+      GstCaps *caps;
+
+      acceptable_caps = gst_pad_get_current_caps (mysrcpad);
+      acceptable_caps = gst_caps_make_writable (acceptable_caps);
+      gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
+          NULL);
+
+      gst_query_parse_caps (query, &filter);
+
+      if (filter)
+        caps =
+            gst_caps_intersect_full (filter, acceptable_caps,
+            GST_CAPS_INTERSECT_FIRST);
+      else
+        caps = gst_caps_ref (acceptable_caps);
+
+      gst_query_set_caps_result (query, caps);
+
+      gst_caps_unref (caps);
+      gst_caps_unref (acceptable_caps);
+
+      return TRUE;
+    }
+    default:
+      return gst_pad_query_default (pad, parent, query);
+  }
+}
+
+static void
+live_switch_push_downstream (gint pts, gint rate)
+{
+  GstBuffer *inbuffer;
+
+  current_rate = rate;
+  gst_pad_push_event (mysinkpad, gst_event_new_reconfigure ());
+
+  inbuffer = gst_buffer_new_and_alloc (48000 * 4 * 2);
+  gst_buffer_memset (inbuffer, 0, 0, 48000 * 4 * 2);
+
+  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
+  GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND;
+  GST_BUFFER_OFFSET (inbuffer) = 0;
+  GST_BUFFER_OFFSET_END (inbuffer) = 47999;
+
+  /* pushing gives away my reference ... */
+  fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+  /* ... but it ends up being collected on the global buffer list */
+}
+
+GST_START_TEST (test_live_switch_downstream)
+{
+  GstElement *audioresample;
+  GstEvent *newseg;
+  GstCaps *caps;
+  GstSegment segment;
+  GList *l;
+  guint i;
+
+  audioresample =
+      setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
+
+  gst_pad_set_query_function (mysinkpad, live_switch_sink_query);
+
+  caps = gst_pad_get_current_caps (mysrcpad);
+  fail_unless (gst_caps_is_fixed (caps));
+
+  fail_unless (gst_element_set_state (audioresample,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  newseg = gst_event_new_segment (&segment);
+  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
+
+  /* buffer is directly passed through */
+  live_switch_push_downstream (0, 48000);
+  fail_unless_equals_int (g_list_length (buffers), 1);
+
+  /* Reconfigure downstream to 40000 Hz */
+  live_switch_push_downstream (1, 40000);
+
+  /* one additional buffer is provided with the new sample rate */
+  fail_unless_equals_int (g_list_length (buffers), 2);
+
+  /* Reconfigure downstream to 50000 Hz */
+  live_switch_push_downstream (2, 50000);
+
+  /* two additional buffers are provided. One is the drained remainder of
+   * the previous sample rate, the second is the buffer with the new sample
+   * rate */
+  fail_unless_equals_int (g_list_length (buffers), 4);
+
+  /* Send EOS to drain the remaining samples */
+  fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
+  fail_unless_equals_int (g_list_length (buffers), 5);
+
+  /* Now test that each buffer has the expected samples. We simply check this
+   * by checking whether the timestamps, durations and sizes are matching */
+  for (l = buffers, i = 0; l; l = l->next, i++) {
+    GstBuffer *buffer = GST_BUFFER (l->data);
+
+    switch (i) {
+      case 0:
+        fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND);
+        fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
+            1 * GST_SECOND);
+        fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2);
+        break;
+      case 1:
+        fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND);
+        fail_unless_equals_int (gst_buffer_get_size (buffer), 39966 * 4 * 2);
+        break;
+      case 2:
+        fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
+                GST_BUFFER_DURATION (buffer), 2 * GST_SECOND,
+                GST_SECOND / 40000 + 1));
+        fail_unless_equals_int (gst_buffer_get_size (buffer), 34 * 4 * 2);
+        break;
+      case 3:
+        fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND);
+        fail_unless_equals_int (gst_buffer_get_size (buffer), 49966 * 4 * 2);
+        break;
+      case 4:
+        fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
+                GST_BUFFER_DURATION (buffer), 3 * GST_SECOND,
+                GST_SECOND / 50000 + 1));
+        fail_unless_equals_int (gst_buffer_get_size (buffer), 33 * 4 * 2);
+        break;
+      default:
+        g_assert_not_reached ();
+        break;
+    }
+
+    gst_buffer_unref (buffer);
+  }
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  cleanup_audioresample (audioresample);
+  gst_caps_unref (caps);
+}
+
+GST_END_TEST;
+
 #ifndef GST_DISABLE_PARSE
 
 static GMainLoop *loop;
@@ -1099,6 +1272,7 @@ audioresample_suite (void)
   tcase_add_test (tc_chain, test_reuse);
   tcase_add_test (tc_chain, test_shutdown);
   tcase_add_test (tc_chain, test_live_switch);
+  tcase_add_test (tc_chain, test_live_switch_downstream);
   tcase_add_test (tc_chain, test_timestamp_drift);
   tcase_add_test (tc_chain, test_fft);