mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-10 08:35:58 +00:00
Code Issues 1 Projects Releases Wiki Activity

tests/check: Add unit test for rtxsend's max_size_time property

Browse source
This commit is contained in:
George Kiagiadakis 2013-11-02 19:56:44 +02:00 committed by Wim Taymans
parent 7d530ab59f
commit 3e818e218b
1 changed files with 44 additions and 9 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

53
tests/check/elements/rtprtx.c
View file

@ -1060,7 +1060,7 @@ GST_END_TEST;
struct GenerateTestBuffersData
{
GstElement *src, *payloader, *sink;
GstElement *src, *capsfilter, *payloader, *sink;
GMutex mutex;
GCond cond;
GList *buffers;
@ -1093,6 +1093,7 @@ static GList *
generate_test_buffers (const gint num_buffers, guint ssrc, guint * payload_type)
{
GstElement *bin;
GstCaps *videotestsrc_caps;
gboolean res;
struct GenerateTestBuffersData data;
@ -1105,15 +1106,28 @@ generate_test_buffers (const gint num_buffers, guint ssrc, guint * payload_type)
bin = gst_pipeline_new (NULL);
data.src = gst_element_factory_make ("videotestsrc", NULL);
data.capsfilter = gst_element_factory_make ("capsfilter", NULL);
data.payloader = gst_element_factory_make ("rtpvrawpay", NULL);
data.sink = gst_element_factory_make ("fakesink", NULL);
/* small frame size will cause vrawpay to generate exactly one rtp packet
* per video frame, which we need for the max-size-time test */
videotestsrc_caps =
gst_caps_from_string
("video/x-raw,format=I420,width=10,height=10,framerate=30/1");
g_object_set (data.src, "do-timestamp", TRUE, NULL);
g_object_set (data.capsfilter, "caps", videotestsrc_caps, NULL);
g_object_set (data.payloader, "seqnum-offset", 1, "ssrc", ssrc, NULL);
g_object_set (data.sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (data.sink, "handoff", (GCallback) fakesink_handoff, &data);
gst_bin_add_many (GST_BIN (bin), data.src, data.payloader, data.sink, NULL);
res = gst_element_link_many (data.src, data.payloader, data.sink, NULL);
gst_caps_unref (videotestsrc_caps);
gst_bin_add_many (GST_BIN (bin), data.src, data.capsfilter, data.payloader,
data.sink, NULL);
res = gst_element_link_many (data.src, data.capsfilter, data.payloader,
data.sink, NULL);
fail_unless_equals_int (res, TRUE);
g_mutex_lock (&data.mutex);
@ -1146,9 +1160,10 @@ create_rtx_event (guint seqnum, guint ssrc, guint payload_type)
"payload-type", G_TYPE_UINT, payload_type, NULL));
}
GST_START_TEST (test_rtxsender_packet_retention)
static void
test_rtxsender_packet_retention (gboolean test_with_time)
{
const gint num_buffers = 10;
const gint num_buffers = test_with_time ? 30 : 10;
const gint half_buffers = num_buffers / 2;
const guint ssrc = 1234567;
const guint rtx_payload_type = 99;
@ -1169,7 +1184,14 @@ GST_START_TEST (test_rtxsender_packet_retention)
/* setup element & pads */
rtxsend = gst_check_setup_element ("rtprtxsend");
g_object_set (rtxsend, "max-size-packets", half_buffers,
/* in both cases we want the rtxsend queue to store 'half_buffers'
* amount of buffers at most. In max-size-packets mode, it's trivial.
* In max-size-time mode, we specify almost half a second, which is
* the equivalent of 15 frames in a 30fps video stream */
g_object_set (rtxsend,
"max-size-packets", test_with_time ? 0 : half_buffers,
"max-size-time", test_with_time ? 499 : 0,
"rtx-payload-type", rtx_payload_type, NULL);
srcpad = gst_check_setup_src_pad (rtxsend, &srctemplate);
@ -1263,8 +1285,8 @@ GST_START_TEST (test_rtxsender_packet_retention)
fail_unless_equals_int (GST_READ_UINT16_BE (gst_rtp_buffer_get_payload (&rtp)), j); /* j == rtx seqnum */
/* open the original packet for this rtx packet and verify timestamps */
res = gst_rtp_buffer_map (GST_BUFFER (g_list_nth_data (in_buffers, j)),
GST_MAP_READ, &orig_rtp);
res = gst_rtp_buffer_map (GST_BUFFER (g_list_nth_data (in_buffers,
j - 1)), GST_MAP_READ, &orig_rtp);
fail_unless_equals_int (res, TRUE);
fail_unless_equals_int (gst_rtp_buffer_get_timestamp (&orig_rtp),
gst_rtp_buffer_get_timestamp (&rtp));
@ -1284,6 +1306,18 @@ GST_START_TEST (test_rtxsender_packet_retention)
gst_check_teardown_element (rtxsend);
}
GST_START_TEST (test_rtxsender_max_size_packets)
{
test_rtxsender_packet_retention (FALSE);
}
GST_END_TEST;
GST_START_TEST (test_rtxsender_max_size_time)
{
test_rtxsender_packet_retention (TRUE);
}
GST_END_TEST;
static Suite *
@ -1299,7 +1333,8 @@ rtprtx_suite (void)
tcase_add_test (tc_chain, test_push_forward_seq);
tcase_add_test (tc_chain, test_drop_one_sender);
tcase_add_test (tc_chain, test_drop_multiple_sender);
tcase_add_test (tc_chain, test_rtxsender_packet_retention);
tcase_add_test (tc_chain, test_rtxsender_max_size_packets);
tcase_add_test (tc_chain, test_rtxsender_max_size_time);
return s;
}