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audiomixer: Add simply synchronization test
This commit is contained in:
Sebastian Dröge
parent
caeaade700
commit
e2e4e8e741
1 changed files with 200 additions and 1 deletions
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@ -2,7 +2,8 @@
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*
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* unit test for audiomixer
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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@ -1257,6 +1258,203 @@ GST_START_TEST (test_flush_start_flush_stop)
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GST_END_TEST;
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static void
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handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
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GstPad * pad, gpointer user_data)
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{
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GList **received_buffers = user_data;
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GST_DEBUG ("got buffer %p", buffer);
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*received_buffers =
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g_list_append (*received_buffers, gst_buffer_ref (buffer));
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}
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GST_START_TEST (test_sync)
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{
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GstSegment segment;
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GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
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GstBus *bus;
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GstPad *sinkpad1, *sinkpad2;
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GstPad *queue1_sinkpad, *queue2_sinkpad;
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GstPad *pad;
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gboolean res;
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GstStateChangeReturn state_res;
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GstFlowReturn ret;
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GstEvent *event;
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GstBuffer *buffer;
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GstCaps *caps;
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GList *received_buffers = NULL, *l;
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gint i;
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GstMapInfo map;
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GST_INFO ("preparing test");
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main_loop = g_main_loop_new (NULL, FALSE);
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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/* just an audiomixer and a fakesink */
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queue1 = gst_element_factory_make ("queue", "queue1");
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queue2 = gst_element_factory_make ("queue", "queue2");
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audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
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g_object_set (audiomixer, "blocksize", 500, NULL);
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sink = gst_element_factory_make ("fakesink", "sink");
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g_object_set (sink, "signal-handoffs", TRUE, NULL);
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g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
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&received_buffers);
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gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
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res = gst_element_link (audiomixer, sink);
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fail_unless (res == TRUE, NULL);
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/* set to playing */
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state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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/* create an unconnected sinkpad in audiomixer, should also automatically activate
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* the pad */
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sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
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fail_if (sinkpad1 == NULL, NULL);
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queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
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pad = gst_element_get_static_pad (queue1, "src");
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fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
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gst_object_unref (pad);
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sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
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fail_if (sinkpad2 == NULL, NULL);
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queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
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pad = gst_element_get_static_pad (queue2, "src");
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fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
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gst_object_unref (pad);
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gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
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gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
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caps = gst_caps_new_simple ("audio/x-raw",
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#if G_BYTE_ORDER == G_BIG_ENDIAN
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"format", G_TYPE_STRING, "S16BE",
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#else
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"format", G_TYPE_STRING, "S16LE",
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#endif
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"layout", G_TYPE_STRING, "interleaved",
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"rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
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gst_pad_set_caps (queue1_sinkpad, caps);
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gst_pad_set_caps (queue2_sinkpad, caps);
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gst_caps_unref (caps);
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/* send segment to audiomixer */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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event = gst_event_new_segment (&segment);
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gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
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gst_pad_send_event (queue2_sinkpad, event);
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/* Push buffers */
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buffer = gst_buffer_new_and_alloc (2000);
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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memset (map.data, 1, map.size);
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gst_buffer_unmap (buffer, &map);
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GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (queue1_sinkpad, buffer);
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ck_assert_int_eq (ret, GST_FLOW_OK);
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buffer = gst_buffer_new_and_alloc (2000);
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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memset (map.data, 1, map.size);
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gst_buffer_unmap (buffer, &map);
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GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (queue1_sinkpad, buffer);
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ck_assert_int_eq (ret, GST_FLOW_OK);
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gst_pad_send_event (queue1_sinkpad, gst_event_new_eos ());
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buffer = gst_buffer_new_and_alloc (2000);
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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memset (map.data, 2, map.size);
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gst_buffer_unmap (buffer, &map);
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GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (queue2_sinkpad, buffer);
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ck_assert_int_eq (ret, GST_FLOW_OK);
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buffer = gst_buffer_new_and_alloc (2000);
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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memset (map.data, 2, map.size);
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gst_buffer_unmap (buffer, &map);
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GST_BUFFER_TIMESTAMP (buffer) = 3 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 1 * GST_SECOND;
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (queue2_sinkpad, buffer);
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ck_assert_int_eq (ret, GST_FLOW_OK);
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gst_pad_send_event (queue2_sinkpad, gst_event_new_eos ());
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/* Collect buffers and messages */
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g_main_loop_run (main_loop);
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/* Here we get once we got EOS, for errors we failed */
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/* Should have 8 * 0.5s buffers */
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fail_unless_equals_int (g_list_length (received_buffers), 8);
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for (i = 0, l = received_buffers; l; l = l->next, i++) {
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buffer = l->data;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
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fail_unless (map.data[0] == 0);
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} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
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fail_unless (map.data[0] == 0);
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} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
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fail_unless (map.data[0] == 1);
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} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
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fail_unless (map.data[0] == 1);
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} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
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fail_unless (map.data[0] == 3);
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} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
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fail_unless (map.data[0] == 3);
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} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
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fail_unless (map.data[0] == 2);
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} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
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fail_unless (map.data[0] == 2);
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} else {
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g_assert_not_reached ();
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}
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gst_buffer_unmap (buffer, &map);
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}
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g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
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gst_element_release_request_pad (audiomixer, sinkpad1);
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gst_object_unref (sinkpad1);
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gst_object_unref (queue1_sinkpad);
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gst_element_release_request_pad (audiomixer, sinkpad2);
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gst_object_unref (sinkpad2);
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gst_object_unref (queue2_sinkpad);
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gst_element_set_state (bin, GST_STATE_NULL);
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gst_bus_remove_signal_watch (bus);
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gst_object_unref (bus);
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gst_object_unref (bin);
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g_main_loop_unref (main_loop);
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}
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GST_END_TEST;
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static Suite *
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audiomixer_suite (void)
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@ -1278,6 +1476,7 @@ audiomixer_suite (void)
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tcase_add_test (tc_chain, test_duration_unknown_overrides);
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tcase_add_test (tc_chain, test_loop);
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tcase_add_test (tc_chain, test_flush_start_flush_stop);
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tcase_add_test (tc_chain, test_sync);
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/* Use a longer timeout */
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#ifdef HAVE_VALGRIND
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