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tests: add a test for srtp elements
https://bugzilla.gnome.org/show_bug.cgi?id=772357
This commit is contained in:
Vincent Penquerc'h
Sebastian Dröge
parent
d125d6b18c
commit
9a2df5dc3b
2 changed files with 232 additions and 0 deletions
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@ -170,6 +170,12 @@ check_hlsdemux_m3u8 =
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check_hlsdemux =
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endif
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if USE_SRTP
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check_srtp = elements/srtp
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else
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check_srtp =
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endif
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if WITH_GST_PLAYER_TESTS
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check_player = libs/player
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else
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@ -299,6 +305,7 @@ check_PROGRAMS = \
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$(check_gl) \
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$(check_hlsdemux_m3u8) \
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$(check_hlsdemux) \
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$(check_srtp) \
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$(check_player) \
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$(EXPERIMENTAL_CHECKS)
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225
tests/check/elements/srtp.c
Normal file
225
tests/check/elements/srtp.c
Normal file
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@ -0,0 +1,225 @@
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/* GStreamer unit tests for the srtp elements
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* Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
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* Copyright (C) 2016 Collabora Ltd <vincent.penquerch@collabora.co.uk>
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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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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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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 ("srtpenc", NULL);
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fail_unless (e != NULL);
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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 ("srtpdec", NULL);
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fail_unless (e != NULL);
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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_play)
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{
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GstElement *source_pipeline, *sink_pipeline;
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GstBus *source_bus;
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GstMessage *msg;
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source_pipeline =
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gst_parse_launch
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("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",
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NULL);
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sink_pipeline =
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gst_parse_launch
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("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",
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NULL);
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fail_unless (gst_element_set_state (source_pipeline,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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fail_unless (gst_element_set_state (sink_pipeline,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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source_bus = gst_pipeline_get_bus (GST_PIPELINE (source_pipeline));
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msg =
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gst_bus_timed_pop_filtered (source_bus, GST_CLOCK_TIME_NONE,
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GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
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fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
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gst_message_unref (msg);
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gst_object_unref (source_bus);
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gst_element_set_state (source_pipeline, GST_STATE_NULL);
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gst_element_set_state (sink_pipeline, GST_STATE_NULL);
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gst_object_unref (source_pipeline);
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gst_object_unref (sink_pipeline);
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}
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GST_END_TEST;
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typedef struct
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{
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guint counter;
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guint start_roc;
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} roc_check_data;
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static guint
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get_roc (GstElement * e)
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{
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const GstStructure *s, *ss;
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const GValue *v;
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guint roc = 0;
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g_object_get (e, "stats", &s, NULL);
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v = gst_structure_get_value (s, "streams");
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fail_unless (v);
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v = gst_value_array_get_value (v, 0);
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ss = gst_value_get_structure (v);
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gst_structure_get_uint (ss, "roc", &roc);
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return roc;
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}
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static GstPadProbeReturn
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roc_check_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
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{
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roc_check_data *data = user_data;
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GstElement *e = GST_PAD_PARENT (pad);
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/* record first roc, then wait for 2^16 packets to pass */
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if (data->counter == 0) {
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data->start_roc = get_roc (e);
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} else if (data->counter == 65536) {
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/* get roc and check it's one more than what we started with */
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fail_unless ((get_roc (e) & 0xffff) == ((data->start_roc + 1) & 0xffff));
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}
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data->counter++;
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return GST_PAD_PROBE_OK;
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}
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static GstCaps *
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request_key (void)
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{
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GstCaps *caps;
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caps =
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gst_caps_from_string
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("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");
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return caps;
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}
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GST_START_TEST (test_roc)
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{
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GstElement *source_pipeline, *sink_pipeline;
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GstElement *srtpenc, *srtpdec;
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GstBus *source_bus, *sink_bus;
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GstMessage *msg;
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GstPad *pad;
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roc_check_data source_roc_check_data, sink_roc_check_data;
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source_pipeline =
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gst_parse_launch
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("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",
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NULL);
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sink_pipeline =
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gst_parse_launch
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("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",
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NULL);
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fail_unless (gst_element_set_state (source_pipeline,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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fail_unless (gst_element_set_state (sink_pipeline,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
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source_bus = gst_pipeline_get_bus (GST_PIPELINE (source_pipeline));
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sink_bus = gst_pipeline_get_bus (GST_PIPELINE (sink_pipeline));
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/* install a pad probe on the srtp elements' source pads */
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srtpenc = gst_bin_get_by_name (GST_BIN (source_pipeline), "enc");
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fail_unless (srtpenc != NULL);
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pad = gst_element_get_static_pad (srtpenc, "rtp_src_0");
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fail_unless (pad != NULL);
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source_roc_check_data.counter = 0;
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gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, roc_check_probe,
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&source_roc_check_data, NULL);
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gst_object_unref (pad);
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gst_object_unref (srtpenc);
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srtpdec = gst_bin_get_by_name (GST_BIN (sink_pipeline), "dec");
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fail_unless (srtpdec != NULL);
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g_signal_connect (srtpdec, "request_key", G_CALLBACK (request_key),
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GINT_TO_POINTER (0));
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pad = gst_element_get_static_pad (srtpdec, "rtp_src");
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sink_roc_check_data.counter = 0;
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gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, roc_check_probe,
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&sink_roc_check_data, NULL);
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fail_unless (pad != NULL);
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gst_object_unref (pad);
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gst_object_unref (srtpdec);
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msg =
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gst_bus_timed_pop_filtered (source_bus, GST_CLOCK_TIME_NONE,
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GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
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fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
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gst_message_unref (msg);
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gst_object_unref (source_bus);
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gst_object_unref (sink_bus);
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gst_element_set_state (source_pipeline, GST_STATE_NULL);
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gst_element_set_state (sink_pipeline, GST_STATE_NULL);
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gst_object_unref (source_pipeline);
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gst_object_unref (sink_pipeline);
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}
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GST_END_TEST;
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static Suite *
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srtp_suite (void)
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{
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Suite *s = suite_create ("srtp");
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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_test (tc_chain, test_create_and_unref);
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tcase_add_test (tc_chain, test_play);
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tcase_add_test (tc_chain, test_roc);
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return s;
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}
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GST_CHECK_MAIN (srtp);
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