tests: Add audiointerleave test to show that queuing works

This tests fails without the queuing patch because incoming buffers are
not delivered before they are needed.

https://bugzilla.gnome.org/show_bug.cgi?id=745768
This commit is contained in:
Olivier Crête 2015-07-02 18:37:28 -04:00
parent 10ef6403b6
commit ce6206388c

View file

@ -34,6 +34,8 @@
#include <gst/audio/audio.h>
#include <gst/audio/audio-enumtypes.h>
#include <gst/check/gstharness.h>
static void
gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
GstCaps * caps, GstFormat format, const gchar * stream_id)
@ -436,11 +438,11 @@ src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
{
gint n = GPOINTER_TO_INT (user_data);
gfloat *data;
gint i;
gsize size;
gint i, num_samples;
GstCaps *caps;
guint64 mask;
GstAudioChannelPosition pos;
GstMapInfo map;
fail_unless (gst_buffer_is_writable (buffer));
@ -469,18 +471,18 @@ src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
gst_pad_set_caps (pad, caps);
gst_caps_unref (caps);
size = 48000 * sizeof (gfloat);
data = g_malloc (size);
for (i = 0; i < 48000; i++)
fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE));
fail_unless (map.size % sizeof (gfloat) == 0);
fail_unless (map.size > 480);
num_samples = map.size / sizeof (gfloat);
data = (gfloat *) map.data;
for (i = 0; i < num_samples; i++)
data[i] = (n % 2 == 0) ? -1.0 : 1.0;
gst_buffer_append_memory (buffer, gst_memory_new_wrapped (0, data,
size, 0, size, data, g_free));
GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_DURATION (buffer) = GST_SECOND;
gst_buffer_unmap (buffer, &map);
}
static void
@ -518,7 +520,7 @@ sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
gst_util_uint64_scale (map.size, GST_SECOND,
48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1);
if (n == 0) {
if (n == 0 || n == 3) {
GstAudioChannelPosition pos[2] =
{ GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
@ -536,23 +538,25 @@ sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
g_assert_not_reached ();
}
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", GST_TYPE_BITMASK, mask, NULL);
ccaps = gst_pad_get_current_caps (pad);
fail_unless (gst_caps_is_equal (caps, ccaps));
gst_caps_unref (ccaps);
gst_caps_unref (caps);
if (pad) {
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
"layout", G_TYPE_STRING, "interleaved",
"channel-mask", GST_TYPE_BITMASK, mask, NULL);
ccaps = gst_pad_get_current_caps (pad);
fail_unless (gst_caps_is_equal (caps, ccaps));
gst_caps_unref (ccaps);
gst_caps_unref (caps);
}
#ifdef HAVE_VALGRIND
if (!(RUNNING_ON_VALGRIND))
#endif
for (i = 0; i < map.size / sizeof (float); i += 2) {
fail_unless_equals_float (data[i], -1.0);
fail_unless_equals_float (data[i + 1], 1.0);
if (n != 3)
fail_unless_equals_float (data[i + 1], 1.0);
}
have_data += map.size;
@ -578,6 +582,9 @@ test_audiointerleave_2ch_pipeline (gboolean interleaved)
src1 = gst_element_factory_make ("fakesrc", "src1");
fail_unless (src1 != NULL);
g_object_set (src1, "num-buffers", 4, NULL);
g_object_set (src1, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src1, "signal-handoffs", TRUE, NULL);
g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
@ -587,6 +594,9 @@ test_audiointerleave_2ch_pipeline (gboolean interleaved)
src2 = gst_element_factory_make ("fakesrc", "src2");
fail_unless (src2 != NULL);
g_object_set (src2, "num-buffers", 4, NULL);
g_object_set (src2, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src2, "signal-handoffs", TRUE, NULL);
g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
@ -675,6 +685,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
src1 = gst_element_factory_make ("fakesrc", "src1");
fail_unless (src1 != NULL);
g_object_set (src1, "num-buffers", 4, NULL);
g_object_set (src1, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src1, "signal-handoffs", TRUE, NULL);
g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src1, "handoff",
@ -684,6 +697,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
src2 = gst_element_factory_make ("fakesrc", "src2");
fail_unless (src2 != NULL);
g_object_set (src2, "num-buffers", 4, NULL);
g_object_set (src2, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src2, "signal-handoffs", TRUE, NULL);
g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src2, "handoff",
@ -764,6 +780,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
fail_unless (src1 != NULL);
g_object_set (src1, "num-buffers", 4, NULL);
g_object_set (src1, "signal-handoffs", TRUE, NULL);
g_object_set (src1, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src1, "handoff",
G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
@ -773,6 +792,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
fail_unless (src2 != NULL);
g_object_set (src2, "num-buffers", 4, NULL);
g_object_set (src2, "signal-handoffs", TRUE, NULL);
g_object_set (src2, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src2, "handoff",
G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
@ -862,6 +884,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
fail_unless (src1 != NULL);
g_object_set (src1, "num-buffers", 4, NULL);
g_object_set (src1, "signal-handoffs", TRUE, NULL);
g_object_set (src1, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src1, "handoff",
G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
@ -871,6 +896,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
fail_unless (src2 != NULL);
g_object_set (src2, "num-buffers", 4, NULL);
g_object_set (src2, "signal-handoffs", TRUE, NULL);
g_object_set (src2, "sizetype", 2,
"sizemax", (int) 48000 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
g_signal_connect (src2, "handoff",
G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
@ -933,6 +961,148 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
GST_END_TEST;
static void
forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type)
{
GstEvent *e;
e = gst_harness_pull_event (hsrc);
fail_unless (GST_EVENT_TYPE (e) == type);
gst_harness_push_event (h, e);
}
GST_START_TEST (test_audiointerleave_2ch_smallbuf)
{
GstElement *audiointerleave;
GstHarness *hsrc;
GstHarness *h;
GstHarness *h2;
GstBuffer *buffer;
GstQuery *q;
gint i;
GstEvent *ev;
GstCaps *ecaps, *caps;
audiointerleave = gst_element_factory_make ("audiointerleave", NULL);
g_object_set (audiointerleave, "latency", GST_SECOND / 2,
"output-buffer-duration", GST_SECOND / 4, NULL);
h = gst_harness_new_with_element (audiointerleave, "sink_0", "src");
gst_harness_use_testclock (h);
h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL);
gst_harness_set_src_caps_str (h2, "audio/x-raw, "
"format=" GST_AUDIO_NE (F32) ", channels=(int)1,"
" layout=interleaved, rate=48000, channel-mask=(bitmask)8");
hsrc = gst_harness_new ("fakesrc");
gst_harness_use_testclock (hsrc);
g_object_set (hsrc->element,
"is-live", TRUE,
"sync", TRUE,
"signal-handoffs", TRUE,
"format", GST_FORMAT_TIME,
"sizetype", 2,
"sizemax", (int) 480 * sizeof (gfloat),
"datarate", (int) 48000 * sizeof (gfloat), NULL);
g_signal_connect (hsrc->element, "handoff",
G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
gst_harness_play (hsrc);
gst_harness_crank_single_clock_wait (hsrc);
forward_check_event (h, hsrc, GST_EVENT_STREAM_START);
forward_check_event (h, hsrc, GST_EVENT_CAPS);
forward_check_event (h, hsrc, GST_EVENT_SEGMENT);
gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
for (i = 0; i < 24; i++) {
gst_harness_crank_single_clock_wait (hsrc);
forward_check_event (h, hsrc, GST_EVENT_CAPS);
gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
}
gst_harness_crank_single_clock_wait (h);
gst_event_unref (gst_harness_pull_event (h)); /* stream-start */
ev = gst_harness_pull_event (h); /* caps */
fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev));
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, 2,
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK, 0x9, NULL);
gst_event_parse_caps (ev, &ecaps);
gst_check_caps_equal (ecaps, caps);
gst_caps_unref (caps);
gst_event_unref (ev);
for (i = 0; i < 24; i++)
gst_harness_crank_single_clock_wait (h);
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
(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);
sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
gst_buffer_unref (buffer);
fail_unless_equals_int (gst_harness_buffers_received (h), 1);
for (i = 0; i < 50; i++) {
gst_harness_crank_single_clock_wait (hsrc);
forward_check_event (h, hsrc, GST_EVENT_CAPS);
gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
}
for (i = 0; i < 25; i++)
gst_harness_crank_single_clock_wait (h);
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
(h->element)), 1000 * GST_MSECOND);
buffer = gst_harness_pull (h);
sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
gst_buffer_unref (buffer);
fail_unless_equals_int (gst_harness_buffers_received (h), 2);
for (i = 0; i < 25; i++) {
gst_harness_crank_single_clock_wait (hsrc);
forward_check_event (h, hsrc, GST_EVENT_CAPS);
gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
}
for (i = 0; i < 25; i++)
gst_harness_crank_single_clock_wait (h);
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
(h->element)), 1250 * GST_MSECOND);
buffer = gst_harness_pull (h);
sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
gst_buffer_unref (buffer);
fail_unless_equals_int (gst_harness_buffers_received (h), 3);
gst_harness_push_event (h, gst_event_new_eos ());
for (i = 0; i < 25; i++)
gst_harness_crank_single_clock_wait (h);
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
(h->element)), 1500 * GST_MSECOND);
buffer = gst_harness_pull (h);
sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
gst_buffer_unref (buffer);
fail_unless_equals_int (gst_harness_buffers_received (h), 4);
gst_harness_teardown (h2);
gst_harness_teardown (h);
gst_harness_teardown (hsrc);
gst_object_unref (audiointerleave);
}
GST_END_TEST;
static Suite *
audiointerleave_suite (void)
{
@ -951,6 +1121,7 @@ audiointerleave_suite (void)
tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos);
tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos);
tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos);
tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf);
return s;
}