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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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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
parent
c2794d1ad0
commit
47e374dbc8
1 changed files with 195 additions and 24 deletions
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@ -34,6 +34,8 @@
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#include <gst/audio/audio.h>
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#include <gst/audio/audio-enumtypes.h>
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#include <gst/check/gstharness.h>
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static void
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gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
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GstCaps * caps, GstFormat format, const gchar * stream_id)
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@ -436,11 +438,11 @@ src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
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{
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gint n = GPOINTER_TO_INT (user_data);
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gfloat *data;
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gint i;
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gsize size;
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gint i, num_samples;
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GstCaps *caps;
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guint64 mask;
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GstAudioChannelPosition pos;
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GstMapInfo map;
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fail_unless (gst_buffer_is_writable (buffer));
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@ -469,18 +471,18 @@ src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
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gst_pad_set_caps (pad, caps);
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gst_caps_unref (caps);
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size = 48000 * sizeof (gfloat);
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data = g_malloc (size);
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for (i = 0; i < 48000; i++)
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fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE));
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fail_unless (map.size % sizeof (gfloat) == 0);
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fail_unless (map.size > 480);
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num_samples = map.size / sizeof (gfloat);
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data = (gfloat *) map.data;
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for (i = 0; i < num_samples; i++)
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data[i] = (n % 2 == 0) ? -1.0 : 1.0;
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gst_buffer_append_memory (buffer, gst_memory_new_wrapped (0, data,
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size, 0, size, data, g_free));
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GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
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GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
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GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
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GST_BUFFER_DURATION (buffer) = GST_SECOND;
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gst_buffer_unmap (buffer, &map);
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}
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static void
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@ -518,7 +520,7 @@ sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
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gst_util_uint64_scale (map.size, GST_SECOND,
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48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1);
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if (n == 0) {
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if (n == 0 || n == 3) {
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GstAudioChannelPosition pos[2] =
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{ GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
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gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
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@ -536,6 +538,7 @@ sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
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g_assert_not_reached ();
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}
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if (pad) {
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
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"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
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@ -546,12 +549,13 @@ sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
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fail_unless (gst_caps_is_equal (caps, ccaps));
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gst_caps_unref (ccaps);
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gst_caps_unref (caps);
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}
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#ifdef HAVE_VALGRIND
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if (!(RUNNING_ON_VALGRIND))
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#endif
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for (i = 0; i < map.size / sizeof (float); i += 2) {
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fail_unless_equals_float (data[i], -1.0);
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if (n != 3)
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fail_unless_equals_float (data[i + 1], 1.0);
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}
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have_data += map.size;
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@ -578,6 +582,9 @@ test_audiointerleave_2ch_pipeline (gboolean interleaved)
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src1 = gst_element_factory_make ("fakesrc", "src1");
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fail_unless (src1 != NULL);
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g_object_set (src1, "num-buffers", 4, NULL);
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g_object_set (src1, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src1, "signal-handoffs", TRUE, NULL);
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g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
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@ -587,6 +594,9 @@ test_audiointerleave_2ch_pipeline (gboolean interleaved)
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src2 = gst_element_factory_make ("fakesrc", "src2");
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fail_unless (src2 != NULL);
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g_object_set (src2, "num-buffers", 4, NULL);
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g_object_set (src2, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src2, "signal-handoffs", TRUE, NULL);
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g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
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@ -675,6 +685,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
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src1 = gst_element_factory_make ("fakesrc", "src1");
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fail_unless (src1 != NULL);
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g_object_set (src1, "num-buffers", 4, NULL);
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g_object_set (src1, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src1, "signal-handoffs", TRUE, NULL);
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g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src1, "handoff",
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@ -684,6 +697,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
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src2 = gst_element_factory_make ("fakesrc", "src2");
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fail_unless (src2 != NULL);
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g_object_set (src2, "num-buffers", 4, NULL);
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g_object_set (src2, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src2, "signal-handoffs", TRUE, NULL);
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g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src2, "handoff",
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@ -764,6 +780,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
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fail_unless (src1 != NULL);
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g_object_set (src1, "num-buffers", 4, NULL);
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g_object_set (src1, "signal-handoffs", TRUE, NULL);
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g_object_set (src1, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src1, "handoff",
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G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
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@ -773,6 +792,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
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fail_unless (src2 != NULL);
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g_object_set (src2, "num-buffers", 4, NULL);
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g_object_set (src2, "signal-handoffs", TRUE, NULL);
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g_object_set (src2, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src2, "handoff",
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G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
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@ -862,6 +884,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
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fail_unless (src1 != NULL);
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g_object_set (src1, "num-buffers", 4, NULL);
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g_object_set (src1, "signal-handoffs", TRUE, NULL);
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g_object_set (src1, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src1, "handoff",
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G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
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@ -871,6 +896,9 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
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fail_unless (src2 != NULL);
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g_object_set (src2, "num-buffers", 4, NULL);
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g_object_set (src2, "signal-handoffs", TRUE, NULL);
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g_object_set (src2, "sizetype", 2,
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"sizemax", (int) 48000 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (src2, "handoff",
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G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
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@ -933,6 +961,148 @@ GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
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GST_END_TEST;
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static void
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forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type)
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{
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GstEvent *e;
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e = gst_harness_pull_event (hsrc);
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fail_unless (GST_EVENT_TYPE (e) == type);
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gst_harness_push_event (h, e);
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}
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GST_START_TEST (test_audiointerleave_2ch_smallbuf)
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{
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GstElement *audiointerleave;
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GstHarness *hsrc;
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GstHarness *h;
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GstHarness *h2;
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GstBuffer *buffer;
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GstQuery *q;
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gint i;
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GstEvent *ev;
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GstCaps *ecaps, *caps;
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audiointerleave = gst_element_factory_make ("audiointerleave", NULL);
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g_object_set (audiointerleave, "latency", GST_SECOND / 2,
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"output-buffer-duration", GST_SECOND / 4, NULL);
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h = gst_harness_new_with_element (audiointerleave, "sink_0", "src");
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gst_harness_use_testclock (h);
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h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL);
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gst_harness_set_src_caps_str (h2, "audio/x-raw, "
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"format=" GST_AUDIO_NE (F32) ", channels=(int)1,"
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" layout=interleaved, rate=48000, channel-mask=(bitmask)8");
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hsrc = gst_harness_new ("fakesrc");
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gst_harness_use_testclock (hsrc);
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g_object_set (hsrc->element,
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"is-live", TRUE,
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"sync", TRUE,
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"signal-handoffs", TRUE,
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"format", GST_FORMAT_TIME,
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"sizetype", 2,
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"sizemax", (int) 480 * sizeof (gfloat),
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"datarate", (int) 48000 * sizeof (gfloat), NULL);
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g_signal_connect (hsrc->element, "handoff",
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G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
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gst_harness_play (hsrc);
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gst_harness_crank_single_clock_wait (hsrc);
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forward_check_event (h, hsrc, GST_EVENT_STREAM_START);
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forward_check_event (h, hsrc, GST_EVENT_CAPS);
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forward_check_event (h, hsrc, GST_EVENT_SEGMENT);
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gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
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for (i = 0; i < 24; i++) {
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gst_harness_crank_single_clock_wait (hsrc);
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forward_check_event (h, hsrc, GST_EVENT_CAPS);
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gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
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}
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gst_harness_crank_single_clock_wait (h);
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gst_event_unref (gst_harness_pull_event (h)); /* stream-start */
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ev = gst_harness_pull_event (h); /* caps */
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fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev));
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caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
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"channels", G_TYPE_INT, 2,
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"layout", G_TYPE_STRING, "interleaved",
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"rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK, 0x9, NULL);
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gst_event_parse_caps (ev, &ecaps);
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gst_check_caps_equal (ecaps, caps);
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gst_caps_unref (caps);
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gst_event_unref (ev);
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for (i = 0; i < 24; i++)
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gst_harness_crank_single_clock_wait (h);
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fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
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(h->element)), 750 * GST_MSECOND);
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/* Check that the queue is really empty */
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q = gst_query_new_drain ();
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gst_pad_peer_query (h->srcpad, q);
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gst_query_unref (q);
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buffer = gst_harness_pull (h);
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sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
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gst_buffer_unref (buffer);
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fail_unless_equals_int (gst_harness_buffers_received (h), 1);
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for (i = 0; i < 50; i++) {
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gst_harness_crank_single_clock_wait (hsrc);
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forward_check_event (h, hsrc, GST_EVENT_CAPS);
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gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
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}
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for (i = 0; i < 25; i++)
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gst_harness_crank_single_clock_wait (h);
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fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
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(h->element)), 1000 * GST_MSECOND);
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buffer = gst_harness_pull (h);
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sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
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gst_buffer_unref (buffer);
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fail_unless_equals_int (gst_harness_buffers_received (h), 2);
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for (i = 0; i < 25; i++) {
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gst_harness_crank_single_clock_wait (hsrc);
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forward_check_event (h, hsrc, GST_EVENT_CAPS);
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gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */
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}
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for (i = 0; i < 25; i++)
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gst_harness_crank_single_clock_wait (h);
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fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
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(h->element)), 1250 * GST_MSECOND);
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buffer = gst_harness_pull (h);
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sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
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gst_buffer_unref (buffer);
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fail_unless_equals_int (gst_harness_buffers_received (h), 3);
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gst_harness_push_event (h, gst_event_new_eos ());
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for (i = 0; i < 25; i++)
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gst_harness_crank_single_clock_wait (h);
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fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
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(h->element)), 1500 * GST_MSECOND);
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buffer = gst_harness_pull (h);
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sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
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gst_buffer_unref (buffer);
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fail_unless_equals_int (gst_harness_buffers_received (h), 4);
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gst_harness_teardown (h2);
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gst_harness_teardown (h);
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gst_harness_teardown (hsrc);
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gst_object_unref (audiointerleave);
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}
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GST_END_TEST;
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static Suite *
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audiointerleave_suite (void)
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{
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@ -951,6 +1121,7 @@ audiointerleave_suite (void)
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tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos);
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tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos);
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tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos);
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tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf);
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return s;
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}
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