test: rtpssrcdemux: Test event forwarding

This the first unit test of this element. It adds a test that verify
that events are forwarded correctly.

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 5d9e74b..87b3ca2 100644 (file)
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -256,6 +256,7 @@ check_rtpmanager = \
        elements/rtpred \
        elements/rtpulpfec \
        elements/rtpfunnel
+       elements/rtpssrcdemux \
 else
 check_rtpmanager =
 endif

diff --git a/tests/check/elements/rtpssrcdemux.c b/tests/check/elements/rtpssrcdemux.c
new file mode 100644 (file)
index 0000000..a8b093b
--- /dev/null
+++ b/tests/check/elements/rtpssrcdemux.c
@@ -0,0 +1,189 @@
+/* GStreamer
+ *
+ * Copyright (C) 2018 Collabora Ltd.
+ *               Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstharness.h>
+
+#define TEST_BUF_CLOCK_RATE 8000
+#define TEST_BUF_PT 0
+#define TEST_BUF_SSRC 0x01BADBAD
+#define TEST_BUF_MS  20
+#define TEST_BUF_DURATION (TEST_BUF_MS * GST_MSECOND)
+#define TEST_BUF_SIZE (64000 * TEST_BUF_MS / 1000)
+#define TEST_RTP_TS_DURATION (TEST_BUF_CLOCK_RATE * TEST_BUF_MS / 1000)
+
+static GstCaps *
+generate_caps (void)
+{
+  return gst_caps_new_simple ("application/x-rtp",
+      "media", G_TYPE_STRING, "audio",
+      "clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
+}
+
+static GstBuffer *
+create_buffer (guint seq_num, guint32 ssrc)
+{
+  GstBuffer *buf;
+  guint8 *payload;
+  guint i;
+  GstClockTime dts = seq_num * TEST_BUF_DURATION;
+  guint32 rtp_ts = seq_num * TEST_RTP_TS_DURATION;
+  GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+  buf = gst_rtp_buffer_new_allocate (TEST_BUF_SIZE, 0, 0);
+  GST_BUFFER_DTS (buf) = dts;
+
+  gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
+  gst_rtp_buffer_set_payload_type (&rtp, TEST_BUF_PT);
+  gst_rtp_buffer_set_seq (&rtp, seq_num);
+  gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
+  gst_rtp_buffer_set_ssrc (&rtp, ssrc);
+
+  payload = gst_rtp_buffer_get_payload (&rtp);
+  for (i = 0; i < TEST_BUF_SIZE; i++)
+    payload[i] = 0xff;
+
+  gst_rtp_buffer_unmap (&rtp);
+
+  return buf;
+}
+
+typedef struct
+{
+  GstHarness *rtp_sink;
+  GstHarness *rtcp_sink;
+  GstHarness *rtp_src;
+  GstHarness *rtcp_src;
+} TestContext;
+
+static void
+rtpssrcdemux_pad_added (G_GNUC_UNUSED GstElement * demux, GstPad * src_pad,
+    TestContext * ctx)
+{
+  GstHarness *h;
+
+  h = gst_harness_new_with_element (ctx->rtp_sink->element, NULL,
+      GST_PAD_NAME (src_pad));
+
+  /* FIXME We should also check that pads have current caps, but this is not
+   * currently the case as both pads are created when the first pad receive a
+   * buffer. If the other pad is not linked, you'll get a pad without caps.
+   * Changing this implies not having both pads on 'on-new-ssrc' which would
+   * break rtpbin assumption. */
+
+  if (g_str_has_prefix (GST_PAD_NAME (src_pad), "src_")) {
+    g_assert (ctx->rtp_src == NULL);
+    ctx->rtp_src = h;
+  } else if (g_str_has_prefix (GST_PAD_NAME (src_pad), "rtcp_src_")) {
+    g_assert (ctx->rtcp_src == NULL);
+    ctx->rtcp_src = h;
+  } else {
+    g_assert_not_reached ();
+  }
+}
+
+GST_START_TEST (test_event_forwarding)
+{
+  TestContext ctx = { NULL, };
+  GstHarness *h;
+  GstEvent *event;
+  GstCaps *caps;
+  GstStructure *s;
+  guint ssrc;
+
+  ctx.rtp_sink = h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink",
+      NULL);
+  g_signal_connect (h->element, "pad_added",
+      G_CALLBACK (rtpssrcdemux_pad_added), &ctx);
+
+  ctx.rtcp_sink = gst_harness_new_with_element (h->element, "rtcp_sink", NULL);
+
+  gst_harness_set_src_caps (h, generate_caps ());
+  gst_harness_push (h, create_buffer (0, TEST_BUF_SSRC));
+
+  g_assert (ctx.rtp_src);
+  g_assert (ctx.rtcp_src);
+
+  gst_harness_push_event (h, gst_event_new_eos ());
+
+  /* We expect stream-start/caps/segment/eos */
+  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 4);
+
+  event = gst_harness_pull_event (ctx.rtp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
+  gst_event_unref (event);
+
+  event = gst_harness_pull_event (ctx.rtp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_CAPS);
+  gst_event_parse_caps (event, &caps);
+  s = gst_caps_get_structure (caps, 0);
+  g_assert (gst_structure_has_field (s, "ssrc"));
+  g_assert (gst_structure_get_uint (s, "ssrc", &ssrc));
+  g_assert_cmpuint (ssrc, ==, TEST_BUF_SSRC);
+  gst_event_unref (event);
+
+  event = gst_harness_pull_event (ctx.rtp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_SEGMENT);
+  gst_event_unref (event);
+
+  event = gst_harness_pull_event (ctx.rtp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
+  gst_event_unref (event);
+
+  /* We pushed on the RTP pad, no events should have reached the RTCP pad */
+  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 0);
+
+  /* push EOS on the rtcp sink pad, to make sure it EOS properly, the harness
+   * will create the missing stream-start */
+  gst_harness_push_event (ctx.rtcp_sink, gst_event_new_eos ());
+
+  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 0);
+  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 2);
+
+  event = gst_harness_pull_event (ctx.rtcp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
+  gst_event_unref (event);
+
+  event = gst_harness_pull_event (ctx.rtcp_src);
+  g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
+  gst_event_unref (event);
+
+  gst_harness_teardown (ctx.rtp_src);
+  gst_harness_teardown (ctx.rtcp_src);
+  gst_harness_teardown (ctx.rtcp_sink);
+  gst_harness_teardown (ctx.rtp_sink);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpssrcdemux_suite (void)
+{
+  Suite *s = suite_create ("rtpssrcdemux");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_event_forwarding);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtpssrcdemux);

diff --git a/tests/check/meson.build b/tests/check/meson.build
index 4b7f3f9..7ba0cb2 100644 (file)
--- a/tests/check/meson.build
+++ b/tests/check/meson.build
@@ -83,6 +83,7 @@ good_tests = [
   [ 'elements/rtpstorage' ],
   [ 'elements/rtpred' ],
   [ 'elements/rtpulpfec' ],
+  [ 'elements/rtpssrcdemux' ],
   [ 'elements/souphttpsrc', not libsoup_dep.found(), [libsoup_dep] ],
   [ 'elements/spectrum' ],
   [ 'elements/shapewipe' ],