tests: add a test for srtp elements

https://bugzilla.gnome.org/show_bug.cgi?id=772357

2 files changed, 232 insertions, 0 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 2c0cdb288..35a4a251d 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -170,6 +170,12 @@ check_hlsdemux_m3u8 =
 check_hlsdemux =
 endif
 
+if USE_SRTP
+check_srtp = elements/srtp
+else
+check_srtp =
+endif
+
 if WITH_GST_PLAYER_TESTS
 check_player = libs/player
 else
@@ -299,6 +305,7 @@ check_PROGRAMS = \
 	$(check_gl) \
 	$(check_hlsdemux_m3u8) \
 	$(check_hlsdemux) \
+	$(check_srtp) \
 	$(check_player) \
 	$(EXPERIMENTAL_CHECKS)
 
diff --git a/tests/check/elements/srtp.c b/tests/check/elements/srtp.c
new file mode 100644
index 000000000..e498afa0e
--- /dev/null
+++ b/tests/check/elements/srtp.c
@@ -0,0 +1,225 @@
+/* GStreamer unit tests for the srtp elements
+ * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 2016 Collabora Ltd <vincent.penquerch@collabora.co.uk>
+ *
+ * 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.
+ */
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#ifdef HAVE_VALGRIND
+# include <valgrind/valgrind.h>
+#endif
+
+#include <gst/check/gstcheck.h>
+
+#include <gst/check/gstharness.h>
+
+GST_START_TEST (test_create_and_unref)
+{
+  GstElement *e;
+
+  e = gst_element_factory_make ("srtpenc", NULL);
+  fail_unless (e != NULL);
+  gst_element_set_state (e, GST_STATE_NULL);
+  gst_object_unref (e);
+
+  e = gst_element_factory_make ("srtpdec", NULL);
+  fail_unless (e != NULL);
+  gst_element_set_state (e, GST_STATE_NULL);
+  gst_object_unref (e);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play)
+{
+  GstElement *source_pipeline, *sink_pipeline;
+  GstBus *source_bus;
+  GstMessage *msg;
+
+  source_pipeline =
+      gst_parse_launch
+      ("audiotestsrc num-buffers=50 ! alawenc ! rtppcmapay ! application/x-rtp, payload=(int)8, ssrc=(uint)1356955624 ! srtpenc name=enc key=012345678901234567890123456789012345678901234567890123456789 ! udpsink port=5004 sync=false",
+      NULL);
+  sink_pipeline =
+      gst_parse_launch
+      ("udpsrc port=5004 caps=\"application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80\" ! srtpdec name=dec ! rtppcmadepay ! alawdec ! fakesink",
+      NULL);
+
+  fail_unless (gst_element_set_state (source_pipeline,
+          GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
+  fail_unless (gst_element_set_state (sink_pipeline,
+          GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
+
+  source_bus = gst_pipeline_get_bus (GST_PIPELINE (source_pipeline));
+
+  msg =
+      gst_bus_timed_pop_filtered (source_bus, GST_CLOCK_TIME_NONE,
+      GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
+  fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
+  gst_message_unref (msg);
+
+  gst_object_unref (source_bus);
+
+  gst_element_set_state (source_pipeline, GST_STATE_NULL);
+  gst_element_set_state (sink_pipeline, GST_STATE_NULL);
+
+  gst_object_unref (source_pipeline);
+  gst_object_unref (sink_pipeline);
+}
+
+GST_END_TEST;
+
+typedef struct
+{
+  guint counter;
+  guint start_roc;
+} roc_check_data;
+
+static guint
+get_roc (GstElement * e)
+{
+  const GstStructure *s, *ss;
+  const GValue *v;
+  guint roc = 0;
+
+  g_object_get (e, "stats", &s, NULL);
+  v = gst_structure_get_value (s, "streams");
+  fail_unless (v);
+  v = gst_value_array_get_value (v, 0);
+  ss = gst_value_get_structure (v);
+  gst_structure_get_uint (ss, "roc", &roc);
+  return roc;
+}
+
+static GstPadProbeReturn
+roc_check_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+  roc_check_data *data = user_data;
+  GstElement *e = GST_PAD_PARENT (pad);
+
+  /* record first roc, then wait for 2^16 packets to pass */
+  if (data->counter == 0) {
+    data->start_roc = get_roc (e);
+  } else if (data->counter == 65536) {
+    /* get roc and check it's one more than what we started with */
+    fail_unless ((get_roc (e) & 0xffff) == ((data->start_roc + 1) & 0xffff));
+  }
+  data->counter++;
+  return GST_PAD_PROBE_OK;
+}
+
+static GstCaps *
+request_key (void)
+{
+  GstCaps *caps;
+
+  caps =
+      gst_caps_from_string
+      ("application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80");
+  return caps;
+}
+
+GST_START_TEST (test_roc)
+{
+  GstElement *source_pipeline, *sink_pipeline;
+  GstElement *srtpenc, *srtpdec;
+  GstBus *source_bus, *sink_bus;
+  GstMessage *msg;
+  GstPad *pad;
+  roc_check_data source_roc_check_data, sink_roc_check_data;
+
+  source_pipeline =
+      gst_parse_launch
+      ("audiotestsrc num-buffers=65555 ! alawenc ! rtppcmapay ! application/x-rtp, payload=(int)8, ssrc=(uint)1356955624 ! srtpenc name=enc key=012345678901234567890123456789012345678901234567890123456789 ! udpsink port=5004 sync=false",
+      NULL);
+  sink_pipeline =
+      gst_parse_launch
+      ("udpsrc port=5004 caps=\"application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80\" ! srtpdec name=dec ! rtppcmadepay ! alawdec ! fakesink",
+      NULL);
+
+  fail_unless (gst_element_set_state (source_pipeline,
+          GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
+  fail_unless (gst_element_set_state (sink_pipeline,
+          GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
+
+  source_bus = gst_pipeline_get_bus (GST_PIPELINE (source_pipeline));
+  sink_bus = gst_pipeline_get_bus (GST_PIPELINE (sink_pipeline));
+
+  /* install a pad probe on the srtp elements' source pads */
+  srtpenc = gst_bin_get_by_name (GST_BIN (source_pipeline), "enc");
+  fail_unless (srtpenc != NULL);
+  pad = gst_element_get_static_pad (srtpenc, "rtp_src_0");
+  fail_unless (pad != NULL);
+  source_roc_check_data.counter = 0;
+  gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, roc_check_probe,
+      &source_roc_check_data, NULL);
+  gst_object_unref (pad);
+  gst_object_unref (srtpenc);
+
+  srtpdec = gst_bin_get_by_name (GST_BIN (sink_pipeline), "dec");
+  fail_unless (srtpdec != NULL);
+  g_signal_connect (srtpdec, "request_key", G_CALLBACK (request_key),
+      GINT_TO_POINTER (0));
+  pad = gst_element_get_static_pad (srtpdec, "rtp_src");
+  sink_roc_check_data.counter = 0;
+  gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, roc_check_probe,
+      &sink_roc_check_data, NULL);
+  fail_unless (pad != NULL);
+  gst_object_unref (pad);
+  gst_object_unref (srtpdec);
+
+  msg =
+      gst_bus_timed_pop_filtered (source_bus, GST_CLOCK_TIME_NONE,
+      GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
+  fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
+  gst_message_unref (msg);
+
+  gst_object_unref (source_bus);
+  gst_object_unref (sink_bus);
+
+  gst_element_set_state (source_pipeline, GST_STATE_NULL);
+  gst_element_set_state (sink_pipeline, GST_STATE_NULL);
+
+  gst_object_unref (source_pipeline);
+  gst_object_unref (sink_pipeline);
+}
+
+GST_END_TEST;
+
+static Suite *
+srtp_suite (void)
+{
+  Suite *s = suite_create ("srtp");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_create_and_unref);
+  tcase_add_test (tc_chain, test_play);
+  tcase_add_test (tc_chain, test_roc);
+
+  return s;
+}
+
+GST_CHECK_MAIN (srtp);