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audioresample: Add unit test for checking for timestamp drifts

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This also checks for perfect timestamping and offsetting.
This commit is contained in:
Sebastian Dröge 2009-08-21 11:51:47 +02:00
parent 1e450f21f8
commit e22c843d0e
1 changed files with 169 additions and 0 deletions
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169
tests/check/elements/audioresample.c
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@ -731,6 +731,174 @@ GST_END_TEST;
#endif
static void
_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
case GST_MESSAGE_WARNING:
g_assert_not_reached ();
break;
case GST_MESSAGE_EOS:
g_main_loop_quit (loop);
break;
default:
break;
}
}
typedef struct
{
guint64 latency;
GstClockTime in_ts;
GstClockTime next_out_ts;
guint64 next_out_off;
guint64 in_buffer_count, out_buffer_count;
} TimestampDriftCtx;
void
fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
TimestampDriftCtx *ctx = user_data;
ctx->out_buffer_count++;
if (ctx->latency == GST_CLOCK_TIME_NONE) {
ctx->latency = 1000 - GST_BUFFER_SIZE (buffer) / 8;
}
/* Check if we have a perfectly timestampped stream */
if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
"expected timestamp %" GST_TIME_FORMAT " got timestamp %"
GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
/* Check if we have a perfectly offsetted stream */
fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
"expected offset end %" G_GUINT64_FORMAT " got offset end %"
G_GUINT64_FORMAT,
GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
GST_BUFFER_OFFSET_END (buffer));
if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
"expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
}
if (ctx->in_buffer_count != ctx->out_buffer_count) {
g_print ("timestamp %" GST_TIME_FORMAT "\n",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
}
if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
&& ctx->in_buffer_count == ctx->out_buffer_count) {
fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
4096),
"expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
GST_SECOND, 4096)),
ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_BUFFER_TIMESTAMP (buffer));
}
ctx->next_out_ts =
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
}
void
identity_handoff_cb (GstElement * object, GstBuffer * buffer,
gpointer user_data)
{
TimestampDriftCtx *ctx = user_data;
ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
ctx->in_buffer_count++;
}
GST_START_TEST (test_timestamp_drift)
{
TimestampDriftCtx ctx =
{ GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
GST_BUFFER_OFFSET_NONE, 0, 0
};
GstElement *pipeline;
GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
*capsfilter2, *fakesink;
GstBus *bus;
GMainLoop *loop;
GstCaps *caps;
pipeline = gst_pipeline_new ("pipeline");
fail_unless (pipeline != NULL);
audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
fail_unless (audiotestsrc != NULL);
g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
"samplesperbuffer", 4000, NULL);
capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
fail_unless (capsfilter1 != NULL);
caps =
gst_caps_from_string
("audio/x-raw-float, channels=1, width=64, rate=16384");
g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
gst_caps_unref (caps);
identity = gst_element_factory_make ("identity", "identity");
fail_unless (identity != NULL);
g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
NULL);
g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
audioresample = gst_element_factory_make ("audioresample", "resample");
fail_unless (audioresample != NULL);
capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
fail_unless (capsfilter2 != NULL);
caps =
gst_caps_from_string
("audio/x-raw-float, channels=1, width=64, rate=4096");
g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
gst_caps_unref (caps);
fakesink = gst_element_factory_make ("fakesink", "sink");
fail_unless (fakesink != NULL);
g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
"signal-handoffs", TRUE, NULL);
g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
audioresample, capsfilter2, fakesink, NULL);
fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
audioresample, capsfilter2, fakesink, NULL));
loop = g_main_loop_new (NULL, FALSE);
bus = gst_element_get_bus (pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_run (loop);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
} GST_END_TEST;
static Suite *
audioresample_suite (void)
{
@ -743,6 +911,7 @@ 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_timestamp_drift);
#ifndef GST_DISABLE_PARSE
tcase_set_timeout (tc_chain, 360);