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srt: Add basic check test of srt[src|sink]
Add some basic tests for the srtsrc and the srtsink and set us up for more advanced tests down the road. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5440>
This commit is contained in:
Jonas K Danielsson
GStreamer Marge Bot
parent
e93298e882
commit
7c4e6442db
192
subprojects/gst-plugins-bad/tests/check/elements/srt.c
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192
subprojects/gst-plugins-bad/tests/check/elements/srt.c
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@ -0,0 +1,192 @@
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/* GStreamer
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* Copyright (C) 2023 Jonas Danielsson <jonas.danielsson@spiideo.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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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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*/
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/* Using GValueArray for stats */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#include "../ext/srt/gstsrt-enums.h"
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static const gchar elements[][8] = { "srtsrc", "srtsink" };
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static void
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check_play (const gchar * src_uri,
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GstSRTConnectionMode src_mode,
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const gchar * sink_uri, GstSRTConnectionMode sink_mode)
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{
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GstHarness *h_src, *h_sink;
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GstStructure *stats;
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gint64 packets_received;
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GstBuffer *in_buf, *out_buf;
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gchar *src_launchline, *sink_launchline;
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GstElement *src_element;
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guint8 data[1316] = { 0 };
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sink_launchline = g_strdup_printf ("srtsink uri=%s", sink_uri);
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h_sink = gst_harness_new_parse (sink_launchline);
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g_free (sink_launchline);
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src_launchline = g_strdup_printf ("srtsrc name=src uri=%s", src_uri);
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h_src = gst_harness_new_parse (src_launchline);
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g_free (src_launchline);
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gst_harness_set_src_caps_str (h_sink, "video/mpegts");
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if (src_mode == GST_SRT_CONNECTION_MODE_LISTENER) {
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fail_unless (gst_element_set_state (h_src->element,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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fail_unless (gst_element_set_state (h_sink->element,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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} else {
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fail_unless (gst_element_set_state (h_sink->element,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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fail_unless (gst_element_set_state (h_src->element,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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}
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in_buf = gst_buffer_new ();
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gst_buffer_append_memory (in_buf,
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gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY,
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data, sizeof (data), 0, sizeof (data), NULL, NULL));
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gst_harness_push (h_sink, in_buf);
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out_buf = gst_harness_pull (h_src);
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src_element = gst_bin_get_by_name (GST_BIN (h_src->element), "src");
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g_object_get (src_element, "stats", &stats, NULL);
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g_assert_cmpstr (gst_structure_get_name (stats), ==,
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"application/x-srt-statistics");
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if (src_mode == GST_SRT_CONNECTION_MODE_CALLER) {
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fail_unless (gst_structure_get_int64 (stats,
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"packets-received", &packets_received));
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g_assert_cmpint (packets_received, ==, 1);
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} else {
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const GValue *array_value;
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const GValue *value;
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GValueArray *callers;
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GstStructure *caller_stats;
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fail_unless ((array_value = gst_structure_get_value (stats, "callers")));
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callers = (GValueArray *) g_value_get_boxed (array_value);
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value = g_value_array_get_nth (callers, 0);
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caller_stats = GST_STRUCTURE (g_value_get_boxed (value));
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fail_unless (gst_structure_get_int64 (caller_stats,
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"packets-received", &packets_received));
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g_assert_cmpint (packets_received, ==, 1);
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}
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gst_element_set_state (h_src->element, GST_STATE_NULL);
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gst_element_set_state (h_sink->element, GST_STATE_NULL);
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gst_buffer_unref (out_buf);
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gst_structure_free (stats);
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gst_object_unref (src_element);
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gst_harness_teardown (h_src);
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gst_harness_teardown (h_sink);
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}
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GST_START_TEST (test_create_and_unref)
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{
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GstElement *e;
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e = gst_element_factory_make (elements[__i__], NULL);
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g_assert_nonnull (e);
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gst_element_set_state (e, GST_STATE_NULL);
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gst_object_unref (e);
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e = gst_element_factory_make (elements[__i__], NULL);
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g_assert_nonnull (e);
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gst_element_set_state (e, GST_STATE_NULL);
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gst_object_unref (e);
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}
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GST_END_TEST;
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GST_START_TEST (test_uri_to_properties)
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{
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GstElement *element;
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gint latency = 0, poll_timeout = 0, mode = 0, pbkeylen = 0;
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guint localport = 0;
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gchar *streamid = NULL, *localaddress = NULL;
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element = gst_element_factory_make (elements[__i__], NULL);
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/* Sets properties to non-default values (make sure this stays in sync) */
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g_object_set (element, "uri", "srt://83.0.2.14:4847?"
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"latency=300" "&mode=listener" "&streamid=the-stream-id"
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"&pbkeylen=32" "&poll-timeout=500", NULL);
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g_object_get (element,
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"latency", &latency, "mode", &mode, "streamid", &streamid,
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"pbkeylen", &pbkeylen, "poll-timeout", &poll_timeout,
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"localport", &localport, "localaddress", &localaddress, NULL);
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/* Make sure these values are in sync with the one from the URI. */
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g_assert_cmpint (latency, ==, 300);
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g_assert_cmpint (mode, ==, 2);
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g_assert_cmpstr (streamid, ==, "the-stream-id");
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g_assert_cmpint (pbkeylen, ==, 32);
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g_assert_cmpint (poll_timeout, ==, 500);
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g_assert_cmpstr (localaddress, ==, "83.0.2.14");
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g_assert_cmpint (localport, ==, 4847);
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g_free (streamid);
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g_free (localaddress);
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gst_object_unref (element);
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}
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GST_END_TEST;
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GST_START_TEST (test_src_caller_sink_listener)
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{
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check_play ("srt://127.0.0.1:3434?mode=caller",
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GST_SRT_CONNECTION_MODE_CALLER,
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"srt://:3434?mode=listener", GST_SRT_CONNECTION_MODE_LISTENER);
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}
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GST_END_TEST;
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GST_START_TEST (test_src_listener_sink_caller)
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{
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check_play ("srt://:4242?mode=listener",
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GST_SRT_CONNECTION_MODE_LISTENER,
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"srt://127.0.0.1:4242?mode=caller", GST_SRT_CONNECTION_MODE_CALLER);
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}
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GST_END_TEST;
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static Suite *
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srt_suite (void)
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{
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Suite *s = suite_create ("srt");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_loop_test (tc_chain, test_create_and_unref, 0,
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G_N_ELEMENTS (elements));
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tcase_add_loop_test (tc_chain, test_uri_to_properties, 0,
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G_N_ELEMENTS (elements));
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tcase_add_test (tc_chain, test_src_caller_sink_listener);
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tcase_add_test (tc_chain, test_src_listener_sink_caller);
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return s;
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}
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GST_CHECK_MAIN (srt);
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@ -69,6 +69,7 @@ base_tests = [
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[['elements/rtpsrc.c'], get_option('rtp').disabled()],
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[['elements/rtpsink.c'], get_option('rtp').disabled()],
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[['elements/sdpdemux.c'], get_option('sdp').disabled(), [gstsdp_dep]],
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[['elements/srt.c'], not srt_dep.found(), [srt_dep]],
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[['elements/srtp.c'], not srtp_dep.found(), [srtp_dep]],
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[['elements/switchbin.c'], get_option('switchbin').disabled()],
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[['elements/videoframe-audiolevel.c'], get_option('videoframe_audiolevel').disabled()],
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