3 * Copyright (C) 2018 Collabora Ltd.
4 * Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
5 * Copyright (C) 2019 Pexip
6 * Author: Havard Graff <havard@pexip.com>
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Library General Public License for more details.
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
21 * Boston, MA 02110-1301, USA.
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include <gst/rtp/gstrtcpbuffer.h>
30 #include <gst/check/gstcheck.h>
31 #include <gst/check/gstharness.h>
34 # include <valgrind/valgrind.h>
36 # define RUNNING_ON_VALGRIND 0
39 #define TEST_BUF_CLOCK_RATE 8000
41 #define TEST_BUF_SSRC 0x01BADBAD
42 #define TEST_BUF_MS 20
43 #define TEST_BUF_DURATION (TEST_BUF_MS * GST_MSECOND)
44 #define TEST_BUF_SIZE (64000 * TEST_BUF_MS / 1000)
45 #define TEST_RTP_TS_DURATION (TEST_BUF_CLOCK_RATE * TEST_BUF_MS / 1000)
50 return gst_caps_new_simple ("application/x-rtp",
51 "media", G_TYPE_STRING, "audio",
52 "clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
56 create_buffer (guint seq_num, guint32 ssrc)
61 GstClockTime dts = seq_num * TEST_BUF_DURATION;
62 guint32 rtp_ts = seq_num * TEST_RTP_TS_DURATION;
63 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
65 buf = gst_rtp_buffer_new_allocate (TEST_BUF_SIZE, 0, 0);
66 GST_BUFFER_DTS (buf) = dts;
68 gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
69 gst_rtp_buffer_set_payload_type (&rtp, TEST_BUF_PT);
70 gst_rtp_buffer_set_seq (&rtp, seq_num);
71 gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
72 gst_rtp_buffer_set_ssrc (&rtp, ssrc);
74 payload = gst_rtp_buffer_get_payload (&rtp);
75 for (i = 0; i < TEST_BUF_SIZE; i++)
78 gst_rtp_buffer_unmap (&rtp);
87 GstHarness *rtcp_sink;
93 rtpssrcdemux_pad_added (G_GNUC_UNUSED GstElement * demux, GstPad * src_pad,
98 h = gst_harness_new_with_element (ctx->rtp_sink->element, NULL,
99 GST_PAD_NAME (src_pad));
101 /* FIXME We should also check that pads have current caps, but this is not
102 * currently the case as both pads are created when the first pad receive a
103 * buffer. If the other pad is not linked, you'll get a pad without caps.
104 * Changing this implies not having both pads on 'on-new-ssrc' which would
105 * break rtpbin assumption. */
107 if (g_str_has_prefix (GST_PAD_NAME (src_pad), "src_")) {
108 g_assert (ctx->rtp_src == NULL);
110 } else if (g_str_has_prefix (GST_PAD_NAME (src_pad), "rtcp_src_")) {
111 g_assert (ctx->rtcp_src == NULL);
114 g_assert_not_reached ();
118 GST_START_TEST (test_event_forwarding)
120 TestContext ctx = { NULL, NULL, NULL, NULL };
127 ctx.rtp_sink = h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink",
129 g_signal_connect (h->element, "pad_added",
130 G_CALLBACK (rtpssrcdemux_pad_added), &ctx);
132 ctx.rtcp_sink = gst_harness_new_with_element (h->element, "rtcp_sink", NULL);
134 gst_harness_set_src_caps (h, generate_caps ());
135 gst_harness_push (h, create_buffer (0, TEST_BUF_SSRC));
137 g_assert (ctx.rtp_src);
138 g_assert (ctx.rtcp_src);
140 gst_harness_push_event (h, gst_event_new_eos ());
142 /* We expect stream-start/caps/segment/eos */
143 g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 4);
145 event = gst_harness_pull_event (ctx.rtp_src);
146 g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
147 gst_event_unref (event);
149 event = gst_harness_pull_event (ctx.rtp_src);
150 g_assert_cmpint (event->type, ==, GST_EVENT_CAPS);
151 gst_event_parse_caps (event, &caps);
152 s = gst_caps_get_structure (caps, 0);
153 g_assert (gst_structure_has_field (s, "ssrc"));
154 g_assert (gst_structure_get_uint (s, "ssrc", &ssrc));
155 g_assert_cmpuint (ssrc, ==, TEST_BUF_SSRC);
156 gst_event_unref (event);
158 event = gst_harness_pull_event (ctx.rtp_src);
159 g_assert_cmpint (event->type, ==, GST_EVENT_SEGMENT);
160 gst_event_unref (event);
162 event = gst_harness_pull_event (ctx.rtp_src);
163 g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
164 gst_event_unref (event);
166 /* We pushed on the RTP pad, no events should have reached the RTCP pad */
167 g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 0);
169 /* push EOS on the rtcp sink pad, to make sure it EOS properly, the harness
170 * will create the missing stream-start */
171 gst_harness_push_event (ctx.rtcp_sink, gst_event_new_eos ());
173 g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 0);
174 g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 1);
176 event = gst_harness_pull_event (ctx.rtcp_src);
177 g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
178 gst_event_unref (event);
180 gst_harness_teardown (ctx.rtp_src);
181 gst_harness_teardown (ctx.rtcp_src);
182 gst_harness_teardown (ctx.rtcp_sink);
183 gst_harness_teardown (ctx.rtp_sink);
196 new_ssrc_pad_cb (G_GNUC_UNUSED GstElement * element, G_GNUC_UNUSED guint ssrc,
197 G_GNUC_UNUSED GstPad * pad, LockTestContext * ctx)
199 g_message ("Signalling ready");
200 g_atomic_int_set (&ctx->ready, 1);
202 g_message ("Waiting no more ready");
203 while (g_atomic_int_get (&ctx->ready))
204 g_usleep (G_USEC_PER_SEC / 100);
206 g_mutex_lock (&ctx->mutex);
207 g_mutex_unlock (&ctx->mutex);
211 push_buffer_func (gpointer user_data)
213 GstHarness *h = user_data;
214 gst_harness_push (h, create_buffer (0, 0xdeadbeef));
218 GST_START_TEST (test_oob_event_locking)
220 GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
224 memset (&ctx, 0, sizeof (LockTestContext));
225 g_mutex_init (&ctx.mutex);
226 g_cond_init (&ctx.cond);
228 gst_harness_set_src_caps_str (h, "application/x-rtp");
229 g_signal_connect (h->element,
230 "new-ssrc-pad", G_CALLBACK (new_ssrc_pad_cb), &ctx);
232 thread = g_thread_new ("streaming-thread", push_buffer_func, h);
234 g_mutex_lock (&ctx.mutex);
236 g_message ("Waiting for ready");
237 while (!g_atomic_int_get (&ctx.ready))
238 g_usleep (G_USEC_PER_SEC / 100);
239 g_message ("Signal no more ready");
240 g_atomic_int_set (&ctx.ready, 0);
242 gst_harness_push_event (h,
243 gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, NULL));
245 g_mutex_unlock (&ctx.mutex);
247 g_thread_join (thread);
248 g_mutex_clear (&ctx.mutex);
249 g_cond_clear (&ctx.cond);
250 gst_harness_teardown (h);
257 new_ssrc_pad_found (GstElement * element, G_GNUC_UNUSED guint ssrc,
258 GstPad * pad, GSList ** src_h)
260 GstHarness *h = gst_harness_new_with_element (element, NULL, NULL);
261 gst_harness_add_element_src_pad (h, pad);
262 *src_h = g_slist_prepend (*src_h, h);
265 GST_START_TEST (test_rtpssrcdemux_max_streams)
267 GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
268 GSList *src_h = NULL;
271 g_object_set (h->element, "max-streams", 64, NULL);
272 gst_harness_set_src_caps_str (h, "application/x-rtp");
273 g_signal_connect (h->element,
274 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, &src_h);
275 gst_harness_play (h);
277 for (i = 0; i < 128; ++i) {
278 fail_unless_equals_int (GST_FLOW_OK,
279 gst_harness_push (h, create_buffer (0, i)));
282 fail_unless_equals_int (g_slist_length (src_h), 64);
283 g_slist_free_full (src_h, (GDestroyNotify) gst_harness_teardown);
284 gst_harness_teardown (h);
290 new_rtcp_ssrc_pad_found (GstElement * element, guint ssrc,
291 G_GNUC_UNUSED GstPad * rtp_pad, GSList ** src_h)
296 name = g_strdup_printf ("rtcp_src_%u", ssrc);
297 h = gst_harness_new_with_element (element, NULL, name);
299 *src_h = g_slist_prepend (*src_h, h);
302 GST_START_TEST (test_rtpssrcdemux_rtcp_app)
305 gst_harness_new_with_padnames ("rtpssrcdemux", "rtcp_sink", NULL);
306 GSList *src_h = NULL;
307 guint8 rtcp_app_pkt[] = { 0x81, 0xcc, 0x00, 0x05, 0x00, 0x00, 0x5d, 0xaf,
308 0x20, 0x20, 0x20, 0x20, 0x21, 0x02, 0x00, 0x0a,
309 0x00, 0x00, 0x5d, 0xaf, 0x00, 0x00, 0x16, 0x03
312 gst_harness_set_src_caps_str (h, "application/x-rtcp");
313 g_signal_connect (h->element,
314 "new-ssrc-pad", (GCallback) new_rtcp_ssrc_pad_found, &src_h);
315 gst_harness_play (h);
317 fail_unless_equals_int (GST_FLOW_OK,
318 gst_harness_push (h, gst_buffer_new_wrapped_full (0, rtcp_app_pkt,
319 sizeof rtcp_app_pkt, 0, sizeof rtcp_app_pkt, NULL, NULL)));
321 fail_unless_equals_int (g_slist_length (src_h), 1);
322 g_slist_free_full (src_h, (GDestroyNotify) gst_harness_teardown);
323 gst_harness_teardown (h);
328 GST_START_TEST (test_rtpssrcdemux_invalid_rtp)
330 GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
335 gst_harness_set_src_caps_str (h, "application/x-rtp");
336 gst_harness_play (h);
338 fail_unless_equals_int (GST_FLOW_OK,
339 gst_harness_push (h, gst_buffer_new_wrapped_full (0, bad_pkt,
340 sizeof bad_pkt, 0, sizeof bad_pkt, NULL, NULL)));
342 gst_harness_teardown (h);
347 GST_START_TEST (test_rtpssrcdemux_invalid_rtcp)
350 gst_harness_new_with_padnames ("rtpssrcdemux", "rtcp_sink", NULL);
355 gst_harness_set_src_caps_str (h, "application/x-rtcp");
356 gst_harness_play (h);
358 fail_unless_equals_int (GST_FLOW_OK,
359 gst_harness_push (h, gst_buffer_new_wrapped_full (0, bad_pkt,
360 sizeof bad_pkt, 0, sizeof bad_pkt, NULL, NULL)));
362 gst_harness_teardown (h);
368 generate_rtcp_sr_buffer (guint ssrc)
371 GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
372 GstRTCPPacket packet;
374 buf = gst_rtcp_buffer_new (1000);
375 fail_unless (gst_rtcp_buffer_map (buf, GST_MAP_READWRITE, &rtcp));
376 fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_SR, &packet));
377 gst_rtcp_packet_sr_set_sender_info (&packet, ssrc, 0, 0, 1, 1);
378 gst_rtcp_buffer_unmap (&rtcp);
389 _simul_ctx_new_ssrc_pad_cb (GstElement * element, guint ssrc,
390 GstPad * rtp_pad, SimulCtx * ctx)
395 gst_harness_add_element_src_pad (ctx->rtp_h, rtp_pad);
397 name = g_strdup_printf ("rtcp_src_%u", ssrc);
398 rtcp_pad = gst_element_get_static_pad (element, name);
399 gst_harness_add_element_src_pad (ctx->rtcp_h, rtcp_pad);
400 gst_object_unref (rtcp_pad);
405 _simul_ctx_push_rtp_buffers (gpointer user_data)
407 SimulCtx *ctx = user_data;
409 gst_harness_set_src_caps_str (ctx->rtp_h, "application/x-rtp");
410 gst_harness_push (ctx->rtp_h, create_buffer (0, 1111));
415 _simul_ctx_push_rtcp_buffers (gpointer user_data)
417 SimulCtx *ctx = user_data;
420 gst_harness_set_src_caps_str (ctx->rtcp_h, "application/x-rtcp");
421 gst_harness_push (ctx->rtcp_h, generate_rtcp_sr_buffer (1111));
425 GST_START_TEST (test_rtp_and_rtcp_arrives_simultaneously)
428 guint repeats = 1000;
429 if (RUNNING_ON_VALGRIND)
432 for (r = 0; r < repeats; r++) {
436 ctx.rtp_h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
438 gst_harness_new_with_element (ctx.rtp_h->element, "rtcp_sink", NULL);
440 g_signal_connect (ctx.rtp_h->element,
441 "new-ssrc-pad", (GCallback) _simul_ctx_new_ssrc_pad_cb, &ctx);
443 t0 = g_thread_new ("push rtp", _simul_ctx_push_rtp_buffers, &ctx);
444 t1 = g_thread_new ("push rtcp", _simul_ctx_push_rtcp_buffers, &ctx);
449 gst_harness_teardown (ctx.rtp_h);
450 gst_harness_teardown (ctx.rtcp_h);
457 rtpssrcdemux_suite (void)
459 Suite *s = suite_create ("rtpssrcdemux");
460 TCase *tc_chain = tcase_create ("general");
462 suite_add_tcase (s, tc_chain);
463 tcase_add_test (tc_chain, test_event_forwarding);
464 tcase_add_test (tc_chain, test_oob_event_locking);
465 tcase_add_test (tc_chain, test_rtpssrcdemux_max_streams);
466 tcase_add_test (tc_chain, test_rtpssrcdemux_rtcp_app);
467 tcase_add_test (tc_chain, test_rtpssrcdemux_invalid_rtp);
468 tcase_add_test (tc_chain, test_rtpssrcdemux_invalid_rtcp);
469 tcase_add_test (tc_chain, test_rtp_and_rtcp_arrives_simultaneously);
474 GST_CHECK_MAIN (rtpssrcdemux);