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audioresample: Add new test that checks for downstream renegotiation
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
GStreamer Merge Bot
parent
71c937b565
commit
8966083178
1 changed files with 174 additions and 0 deletions
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@ -570,6 +570,179 @@ GST_START_TEST (test_live_switch)
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GST_END_TEST;
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static gint current_rate = 0;
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static gboolean
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live_switch_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
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{
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_ACCEPT_CAPS:{
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GstCaps *acceptable_caps;
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GstCaps *caps;
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acceptable_caps = gst_pad_get_current_caps (mysrcpad);
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acceptable_caps = gst_caps_make_writable (acceptable_caps);
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gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
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NULL);
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gst_query_parse_accept_caps (query, &caps);
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gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps,
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acceptable_caps));
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gst_caps_unref (acceptable_caps);
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return TRUE;
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}
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case GST_QUERY_CAPS:{
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GstCaps *acceptable_caps;
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GstCaps *filter;
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GstCaps *caps;
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acceptable_caps = gst_pad_get_current_caps (mysrcpad);
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acceptable_caps = gst_caps_make_writable (acceptable_caps);
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gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
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NULL);
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gst_query_parse_caps (query, &filter);
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if (filter)
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caps =
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gst_caps_intersect_full (filter, acceptable_caps,
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GST_CAPS_INTERSECT_FIRST);
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else
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caps = gst_caps_ref (acceptable_caps);
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gst_query_set_caps_result (query, caps);
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gst_caps_unref (caps);
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gst_caps_unref (acceptable_caps);
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return TRUE;
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}
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default:
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return gst_pad_query_default (pad, parent, query);
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}
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}
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static void
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live_switch_push_downstream (gint pts, gint rate)
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{
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GstBuffer *inbuffer;
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current_rate = rate;
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gst_pad_push_event (mysinkpad, gst_event_new_reconfigure ());
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inbuffer = gst_buffer_new_and_alloc (48000 * 4 * 2);
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gst_buffer_memset (inbuffer, 0, 0, 48000 * 4 * 2);
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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GST_BUFFER_OFFSET_END (inbuffer) = 47999;
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/* pushing gives away my reference ... */
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fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
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/* ... but it ends up being collected on the global buffer list */
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}
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GST_START_TEST (test_live_switch_downstream)
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{
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GstElement *audioresample;
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GstEvent *newseg;
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GstCaps *caps;
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GstSegment segment;
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GList *l;
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guint i;
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audioresample =
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setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
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gst_pad_set_query_function (mysinkpad, live_switch_sink_query);
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caps = gst_pad_get_current_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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gst_segment_init (&segment, GST_FORMAT_TIME);
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newseg = gst_event_new_segment (&segment);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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/* buffer is directly passed through */
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live_switch_push_downstream (0, 48000);
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fail_unless_equals_int (g_list_length (buffers), 1);
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/* Reconfigure downstream to 40000 Hz */
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live_switch_push_downstream (1, 40000);
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/* one additional buffer is provided with the new sample rate */
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fail_unless_equals_int (g_list_length (buffers), 2);
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/* Reconfigure downstream to 50000 Hz */
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live_switch_push_downstream (2, 50000);
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/* two additional buffers are provided. One is the drained remainder of
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* the previous sample rate, the second is the buffer with the new sample
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* rate */
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fail_unless_equals_int (g_list_length (buffers), 4);
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/* Send EOS to drain the remaining samples */
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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fail_unless_equals_int (g_list_length (buffers), 5);
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/* Now test that each buffer has the expected samples. We simply check this
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* by checking whether the timestamps, durations and sizes are matching */
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for (l = buffers, i = 0; l; l = l->next, i++) {
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GstBuffer *buffer = GST_BUFFER (l->data);
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switch (i) {
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case 0:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND);
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fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
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1 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2);
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break;
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case 1:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 39966 * 4 * 2);
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break;
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case 2:
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fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
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GST_BUFFER_DURATION (buffer), 2 * GST_SECOND,
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GST_SECOND / 40000 + 1));
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fail_unless_equals_int (gst_buffer_get_size (buffer), 34 * 4 * 2);
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break;
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case 3:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 49966 * 4 * 2);
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break;
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case 4:
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fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
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GST_BUFFER_DURATION (buffer), 3 * GST_SECOND,
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GST_SECOND / 50000 + 1));
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fail_unless_equals_int (gst_buffer_get_size (buffer), 33 * 4 * 2);
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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gst_buffer_unref (buffer);
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}
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g_list_free (buffers);
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buffers = NULL;
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cleanup_audioresample (audioresample);
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gst_caps_unref (caps);
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}
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GST_END_TEST;
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#ifndef GST_DISABLE_PARSE
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static GMainLoop *loop;
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@ -1099,6 +1272,7 @@ audioresample_suite (void)
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tcase_add_test (tc_chain, test_reuse);
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tcase_add_test (tc_chain, test_shutdown);
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tcase_add_test (tc_chain, test_live_switch);
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tcase_add_test (tc_chain, test_live_switch_downstream);
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tcase_add_test (tc_chain, test_timestamp_drift);
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tcase_add_test (tc_chain, test_fft);
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