tests: add a test for srtp elements

https://bugzilla.gnome.org/show_bug.cgi?id=772357
This commit is contained in:
Vincent Penquerc'h 2016-10-04 16:59:09 +01:00 committed by Sebastian Dröge
parent d125d6b18c
commit 9a2df5dc3b
2 changed files with 232 additions and 0 deletions

View file

@ -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)

225
tests/check/elements/srtp.c Normal file
View file

@ -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);