mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 08:46:40 +00:00
Code Issues 1 Projects Releases Wiki Activity

srtp: add unit tests

Browse source 
Enable unit tests in meson.build
Add test_play_key_error to check the stats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
This commit is contained in:
Stéphane Cerveau 2022-03-28 17:06:46 +02:00 committed by GStreamer Marge Bot
parent c77d07752a
commit 12776ba0fd
2 changed files with 60 additions and 15 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

74
subprojects/gst-plugins-bad/tests/check/elements/srtp.c
View file

@ -20,11 +20,11 @@
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#include "config.h"
#endif
#ifdef HAVE_VALGRIND
# include <valgrind/valgrind.h>
#include <valgrind/valgrind.h>
#endif
#include <gst/check/gstcheck.h>
@ -48,20 +48,35 @@ GST_START_TEST (test_create_and_unref)
GST_END_TEST;
GST_START_TEST (test_play)
static void
check_play (const gchar * encode_key, const gchar * decode_key,
guint buffer_count, guint expected_recv_count,
guint expected_recv_drop_count)
{
GstElement *source_pipeline, *sink_pipeline;
GstBus *source_bus;
GstMessage *msg;
GstStructure *stats;
guint recv_count = 0;
guint drop_count = 0;
GstElement *srtp_dec;
guint port = 5004;
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);
gchar *source_pipeline_desc = g_strdup_printf ("audiotestsrc num-buffers=%d \
! alawenc ! rtppcmapay ! application/x-rtp, payload=(int)8, ssrc=(uint)1356955624 \
! srtpenc name=enc key=%s ! udpsink port=%d sync=false host=127.0.0.1", buffer_count, encode_key, port);
gchar *sink_pipeline_desc =
g_strdup_printf ("udpsrc port=%d caps=\"application/x-srtp, \
payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)%s, 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", port, decode_key);
source_pipeline = gst_parse_launch (source_pipeline_desc, NULL);
sink_pipeline = gst_parse_launch (sink_pipeline_desc, NULL);
g_free (source_pipeline_desc);
g_free (sink_pipeline_desc);
fail_unless (gst_element_set_state (source_pipeline,
GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE);
@ -76,6 +91,17 @@ GST_START_TEST (test_play)
fail_unless (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS);
gst_message_unref (msg);
// Wait 1s that all the buffers reached the sink pipeline entirely
g_usleep (G_USEC_PER_SEC * 1);
srtp_dec = gst_bin_get_by_name (GST_BIN (sink_pipeline), "dec");
g_object_get (srtp_dec, "stats", &stats, NULL);
gst_structure_get_uint (stats, "recv-count", &recv_count);
fail_unless (recv_count <= expected_recv_count);
gst_structure_get_uint (stats, "recv-drop-count", &drop_count);
fail_unless (drop_count <= expected_recv_drop_count);
gst_object_unref (srtp_dec);
gst_structure_free (stats);
gst_object_unref (source_bus);
gst_element_set_state (source_pipeline, GST_STATE_NULL);
@ -85,8 +111,23 @@ GST_START_TEST (test_play)
gst_object_unref (sink_pipeline);
}
GST_START_TEST (test_play)
{
check_play ("012345678901234567890123456789012345678901234567890123456789",
"012345678901234567890123456789012345678901234567890123456789", 50, 50,
0);
}
GST_END_TEST;
GST_START_TEST (test_play_key_error)
{
check_play ("012345678901234567890123456789012345678901234567890123456789",
"000000000000000000000000000000000000000000000000000000000000", 50, 50,
50);
}
GST_END_TEST;
typedef struct
{
guint counter;
@ -96,16 +137,18 @@ typedef struct
static guint
get_roc (GstElement * e)
{
const GstStructure *s, *ss;
GstStructure *stats;
const GstStructure *ss;
const GValue *v;
guint roc = 0;
g_object_get (e, "stats", &s, NULL);
v = gst_structure_get_value (s, "streams");
g_object_get (e, "stats", &stats, NULL);
v = gst_structure_get_value (stats, "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);
gst_structure_free (stats);
return roc;
}
@ -151,7 +194,7 @@ GST_START_TEST (test_roc)
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",
("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 host=127.0.0.1",
NULL);
sink_pipeline =
gst_parse_launch
@ -352,6 +395,7 @@ srtp_suite (void)
tcase_add_test (tc_chain, test_create_and_unref);
tcase_add_test (tc_chain, test_play);
tcase_add_test (tc_chain, test_roc);
tcase_add_test (tc_chain, test_play_key_error);
#ifdef HAVE_SRTP2
tcase_add_test (tc_chain, test_simple_mki);
tcase_add_test (tc_chain, test_srtpdec_multiple_mki);

1
subprojects/gst-plugins-bad/tests/check/meson.build
View file

@ -64,6 +64,7 @@ base_tests = [
[['elements/rtponviftimestamp.c']],
[['elements/rtpsrc.c']],
[['elements/rtpsink.c']],
[['elements/srtp.c']],
[['elements/switchbin.c']],
[['elements/videoframe-audiolevel.c']],
[['elements/viewfinderbin.c']],