Commit c596bdda authored by Nicolas Dufresne's avatar Nicolas Dufresne

test: rtpssrcdemux: Test event forwarding

This the first unit test of this element. It adds a test that verify
that events are forwarded correctly.
parent e9495c55
......@@ -256,6 +256,7 @@ check_rtpmanager = \
elements/rtpred \
elements/rtpulpfec \
elements/rtpfunnel
elements/rtpssrcdemux \
else
check_rtpmanager =
endif
......
/* GStreamer
*
* Copyright (C) 2018 Collabora Ltd.
* Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* 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.
*/
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#define TEST_BUF_CLOCK_RATE 8000
#define TEST_BUF_PT 0
#define TEST_BUF_SSRC 0x01BADBAD
#define TEST_BUF_MS 20
#define TEST_BUF_DURATION (TEST_BUF_MS * GST_MSECOND)
#define TEST_BUF_SIZE (64000 * TEST_BUF_MS / 1000)
#define TEST_RTP_TS_DURATION (TEST_BUF_CLOCK_RATE * TEST_BUF_MS / 1000)
static GstCaps *
generate_caps (void)
{
return gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
}
static GstBuffer *
create_buffer (guint seq_num, guint32 ssrc)
{
GstBuffer *buf;
guint8 *payload;
guint i;
GstClockTime dts = seq_num * TEST_BUF_DURATION;
guint32 rtp_ts = seq_num * TEST_RTP_TS_DURATION;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
buf = gst_rtp_buffer_new_allocate (TEST_BUF_SIZE, 0, 0);
GST_BUFFER_DTS (buf) = dts;
gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, TEST_BUF_PT);
gst_rtp_buffer_set_seq (&rtp, seq_num);
gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
gst_rtp_buffer_set_ssrc (&rtp, ssrc);
payload = gst_rtp_buffer_get_payload (&rtp);
for (i = 0; i < TEST_BUF_SIZE; i++)
payload[i] = 0xff;
gst_rtp_buffer_unmap (&rtp);
return buf;
}
typedef struct
{
GstHarness *rtp_sink;
GstHarness *rtcp_sink;
GstHarness *rtp_src;
GstHarness *rtcp_src;
} TestContext;
static void
rtpssrcdemux_pad_added (G_GNUC_UNUSED GstElement * demux, GstPad * src_pad,
TestContext * ctx)
{
GstHarness *h;
h = gst_harness_new_with_element (ctx->rtp_sink->element, NULL,
GST_PAD_NAME (src_pad));
/* FIXME We should also check that pads have current caps, but this is not
* currently the case as both pads are created when the first pad receive a
* buffer. If the other pad is not linked, you'll get a pad without caps.
* Changing this implies not having both pads on 'on-new-ssrc' which would
* break rtpbin assumption. */
if (g_str_has_prefix (GST_PAD_NAME (src_pad), "src_")) {
g_assert (ctx->rtp_src == NULL);
ctx->rtp_src = h;
} else if (g_str_has_prefix (GST_PAD_NAME (src_pad), "rtcp_src_")) {
g_assert (ctx->rtcp_src == NULL);
ctx->rtcp_src = h;
} else {
g_assert_not_reached ();
}
}
GST_START_TEST (test_event_forwarding)
{
TestContext ctx = { NULL, };
GstHarness *h;
GstEvent *event;
GstCaps *caps;
GstStructure *s;
guint ssrc;
ctx.rtp_sink = h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink",
NULL);
g_signal_connect (h->element, "pad_added",
G_CALLBACK (rtpssrcdemux_pad_added), &ctx);
ctx.rtcp_sink = gst_harness_new_with_element (h->element, "rtcp_sink", NULL);
gst_harness_set_src_caps (h, generate_caps ());
gst_harness_push (h, create_buffer (0, TEST_BUF_SSRC));
g_assert (ctx.rtp_src);
g_assert (ctx.rtcp_src);
gst_harness_push_event (h, gst_event_new_eos ());
/* We expect stream-start/caps/segment/eos */
g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 4);
event = gst_harness_pull_event (ctx.rtp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
gst_event_unref (event);
event = gst_harness_pull_event (ctx.rtp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_CAPS);
gst_event_parse_caps (event, &caps);
s = gst_caps_get_structure (caps, 0);
g_assert (gst_structure_has_field (s, "ssrc"));
g_assert (gst_structure_get_uint (s, "ssrc", &ssrc));
g_assert_cmpuint (ssrc, ==, TEST_BUF_SSRC);
gst_event_unref (event);
event = gst_harness_pull_event (ctx.rtp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_SEGMENT);
gst_event_unref (event);
event = gst_harness_pull_event (ctx.rtp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
gst_event_unref (event);
/* We pushed on the RTP pad, no events should have reached the RTCP pad */
g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 0);
/* push EOS on the rtcp sink pad, to make sure it EOS properly, the harness
* will create the missing stream-start */
gst_harness_push_event (ctx.rtcp_sink, gst_event_new_eos ());
g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 0);
g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 2);
event = gst_harness_pull_event (ctx.rtcp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
gst_event_unref (event);
event = gst_harness_pull_event (ctx.rtcp_src);
g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
gst_event_unref (event);
gst_harness_teardown (ctx.rtp_src);
gst_harness_teardown (ctx.rtcp_src);
gst_harness_teardown (ctx.rtcp_sink);
gst_harness_teardown (ctx.rtp_sink);
}
GST_END_TEST;
static Suite *
rtpssrcdemux_suite (void)
{
Suite *s = suite_create ("rtpssrcdemux");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_event_forwarding);
return s;
}
GST_CHECK_MAIN (rtpssrcdemux);
......@@ -83,6 +83,7 @@ good_tests = [
[ 'elements/rtpstorage' ],
[ 'elements/rtpred' ],
[ 'elements/rtpulpfec' ],
[ 'elements/rtpssrcdemux' ],
[ 'elements/souphttpsrc', not libsoup_dep.found(), [libsoup_dep] ],
[ 'elements/spectrum' ],
[ 'elements/shapewipe' ],
......
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