mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-03 13:16:04 +00:00
Code Issues 1 Projects Releases Wiki Activity

audioresample: Add new test that checks for downstream renegotiation

Browse source 
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>
This commit is contained in:
Sebastian Dröge 2020-05-27 19:59:56 +03:00 committed by GStreamer Merge Bot
parent 71c937b565
commit 8966083178
1 changed files with 174 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

174
tests/check/elements/audioresample.c
View file

@ -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);