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tests: audiointerleave: Remove drain with manual clock

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Now that the queries go onto the queue, you may need to pull the crank
in order for them to be processed, making this test difficult.
This commit is contained in:
Olivier Crête 2017-05-23 00:52:27 +02:00
parent 930c3cea40
commit e537f3576a
1 changed files with 0 additions and 6 deletions
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6
tests/check/elements/audiointerleave.c
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@ -980,7 +980,6 @@ GST_START_TEST (test_audiointerleave_2ch_smallbuf)
GstHarness *h; GstHarness *h;
GstHarness *h2; GstHarness *h2;
GstBuffer *buffer; GstBuffer *buffer;
GstQuery *q;
gint i; gint i;
GstEvent *ev; GstEvent *ev;
GstCaps *ecaps, *caps; GstCaps *ecaps, *caps;
@ -1050,11 +1049,6 @@ GST_START_TEST (test_audiointerleave_2ch_smallbuf)
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
(h->element)), 750 * GST_MSECOND); (h->element)), 750 * GST_MSECOND);
/* Check that the queue is really empty */
q = gst_query_new_drain ();
gst_pad_peer_query (h->srcpad, q);
gst_query_unref (q);
buffer = gst_harness_pull (h); buffer = gst_harness_pull (h);
sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
gst_buffer_unref (buffer); gst_buffer_unref (buffer);