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tests: audiomixer: add unit test for proper segment.base handling

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As adjusted by gst_pad_set_offset(), or when doing segment seeks
or looping for example. See previous audiomixer commit.
This commit is contained in:
Tim-Philipp Müller 2015-02-05 15:23:04 +00:00
parent 6d6c693254
commit e54829aa4f
1 changed files with 65 additions and 0 deletions
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65
tests/check/elements/audiomixer.c
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@ -1774,6 +1774,68 @@ GST_START_TEST (test_sync_unaligned)
GST_END_TEST;
#ifndef GST_DISABLE_PARSE
GST_START_TEST (test_segment_base_handling)
{
GstElement *pipeline, *sink, *mix, *src1, *src2;
GstPad *srcpad, *sinkpad;
GstClockTime end_time;
GstSample *last_sample = NULL;
GstSample *sample;
GstBuffer *buf;
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
"channels", G_TYPE_INT, 2, NULL);
pipeline = gst_pipeline_new ("pipeline");
mix = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("appsink", "sink");
g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
gst_caps_unref (caps);
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
src2 = gst_element_factory_make ("audiotestsrc", "src2");
g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL);
fail_unless (gst_element_link (mix, sink));
srcpad = gst_element_get_static_pad (src1, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_1");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
srcpad = gst_element_get_static_pad (src2, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_2");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
do {
g_signal_emit_by_name (sink, "pull-sample", &sample);
if (sample == NULL)
break;
if (last_sample)
gst_sample_unref (last_sample);
last_sample = sample;
} while (TRUE);
buf = gst_sample_get_buffer (last_sample);
end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
gst_sample_unref (last_sample);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
#endif
static Suite *
audiomixer_suite (void)
{
@ -1797,6 +1859,9 @@ audiomixer_suite (void)
tcase_add_test (tc_chain, test_sync);
tcase_add_test (tc_chain, test_sync_discont);
tcase_add_test (tc_chain, test_sync_unaligned);
#ifndef GST_DISABLE_PARSE
tcase_add_test (tc_chain, test_segment_base_handling);
#endif
/* Use a longer timeout */
#ifdef HAVE_VALGRIND