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synced 2024-12-18 14:26:43 +00:00
audioresample: Add unit test for checking for timestamp drifts
This also checks for perfect timestamping and offsetting.
This commit is contained in:
Sebastian Dröge
parent
1e450f21f8
commit
e22c843d0e
1 changed files with 169 additions and 0 deletions
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@ -731,6 +731,174 @@ GST_END_TEST;
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#endif
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#endif
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static void
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_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
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{
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GMainLoop *loop = user_data;
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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case GST_MESSAGE_WARNING:
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g_assert_not_reached ();
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break;
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case GST_MESSAGE_EOS:
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g_main_loop_quit (loop);
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break;
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default:
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break;
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}
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}
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typedef struct
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{
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guint64 latency;
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GstClockTime in_ts;
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GstClockTime next_out_ts;
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guint64 next_out_off;
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guint64 in_buffer_count, out_buffer_count;
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} TimestampDriftCtx;
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void
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fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
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gpointer user_data)
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{
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TimestampDriftCtx *ctx = user_data;
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ctx->out_buffer_count++;
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if (ctx->latency == GST_CLOCK_TIME_NONE) {
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ctx->latency = 1000 - GST_BUFFER_SIZE (buffer) / 8;
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}
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/* Check if we have a perfectly timestampped stream */
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if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
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fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
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"expected timestamp %" GST_TIME_FORMAT " got timestamp %"
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GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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/* Check if we have a perfectly offsetted stream */
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fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
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GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
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"expected offset end %" G_GUINT64_FORMAT " got offset end %"
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G_GUINT64_FORMAT,
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GST_BUFFER_OFFSET (buffer) + GST_BUFFER_SIZE (buffer) / 8,
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GST_BUFFER_OFFSET_END (buffer));
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if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
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fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
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"expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
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ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
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}
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if (ctx->in_buffer_count != ctx->out_buffer_count) {
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g_print ("timestamp %" GST_TIME_FORMAT "\n",
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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}
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if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
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&& ctx->in_buffer_count == ctx->out_buffer_count) {
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fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
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ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
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4096),
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"expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
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") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
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GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
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GST_SECOND, 4096)),
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ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
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4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
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GST_BUFFER_TIMESTAMP (buffer));
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}
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ctx->next_out_ts =
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GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
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ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
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}
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void
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identity_handoff_cb (GstElement * object, GstBuffer * buffer,
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gpointer user_data)
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{
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TimestampDriftCtx *ctx = user_data;
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ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
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ctx->in_buffer_count++;
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}
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GST_START_TEST (test_timestamp_drift)
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{
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TimestampDriftCtx ctx =
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{ GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
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GST_BUFFER_OFFSET_NONE, 0, 0
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};
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GstElement *pipeline;
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GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
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*capsfilter2, *fakesink;
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GstBus *bus;
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GMainLoop *loop;
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GstCaps *caps;
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pipeline = gst_pipeline_new ("pipeline");
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fail_unless (pipeline != NULL);
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audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
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fail_unless (audiotestsrc != NULL);
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g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
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"samplesperbuffer", 4000, NULL);
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capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
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fail_unless (capsfilter1 != NULL);
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caps =
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gst_caps_from_string
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("audio/x-raw-float, channels=1, width=64, rate=16384");
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g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
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gst_caps_unref (caps);
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identity = gst_element_factory_make ("identity", "identity");
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fail_unless (identity != NULL);
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g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
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NULL);
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g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
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audioresample = gst_element_factory_make ("audioresample", "resample");
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fail_unless (audioresample != NULL);
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capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
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fail_unless (capsfilter2 != NULL);
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caps =
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gst_caps_from_string
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("audio/x-raw-float, channels=1, width=64, rate=4096");
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g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
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gst_caps_unref (caps);
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fakesink = gst_element_factory_make ("fakesink", "sink");
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fail_unless (fakesink != NULL);
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g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
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"signal-handoffs", TRUE, NULL);
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g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
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gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
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audioresample, capsfilter2, fakesink, NULL);
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fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
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audioresample, capsfilter2, fakesink, NULL));
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loop = g_main_loop_new (NULL, FALSE);
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bus = gst_element_get_bus (pipeline);
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gst_bus_add_signal_watch (bus);
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g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
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fail_unless (gst_element_set_state (pipeline,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
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g_main_loop_run (loop);
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fail_unless (gst_element_set_state (pipeline,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
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g_main_loop_unref (loop);
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gst_object_unref (pipeline);
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} GST_END_TEST;
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static Suite *
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static Suite *
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audioresample_suite (void)
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audioresample_suite (void)
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{
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{
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@ -743,6 +911,7 @@ audioresample_suite (void)
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tcase_add_test (tc_chain, test_reuse);
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tcase_add_test (tc_chain, test_reuse);
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tcase_add_test (tc_chain, test_shutdown);
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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);
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tcase_add_test (tc_chain, test_timestamp_drift);
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#ifndef GST_DISABLE_PARSE
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#ifndef GST_DISABLE_PARSE
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tcase_set_timeout (tc_chain, 360);
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tcase_set_timeout (tc_chain, 360);
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