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webrtcbin tests: Add test for intersection src pad caps

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This checks that the codec preferences are intersected also with what
the src pad can handle.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
This commit is contained in:
Olivier Crête 2021-05-06 13:52:32 -04:00
parent cc556452ce
commit 78d2d6cf6f
1 changed files with 113 additions and 0 deletions
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113
tests/check/elements/webrtcbin.c
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@ -738,6 +738,7 @@ test_webrtc_wait_for_ice_connection (struct test_webrtc *t,
g_mutex_unlock (&t->lock);
}
#endif
static void
_pad_added_fakesink (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
@ -3999,6 +4000,117 @@ GST_START_TEST (test_codec_preferences_negotiation_sinkpad)
GST_END_TEST;
static void
add_audio_test_src_harness (GstHarness * h)
{
#define L16_CAPS "application/x-rtp, payload=11, media=audio," \
" encoding-name=L16, clock-rate=44100, ssrc=(uint)3484078952"
GstCaps *caps = gst_caps_from_string (L16_CAPS);
gst_harness_set_src_caps (h, caps);
gst_harness_add_src_parse (h, "audiotestsrc is-live=true ! rtpL16pay ! "
L16_CAPS " ! identity", TRUE);
}
static void
_pad_added_harness (struct test_webrtc *t, GstElement * element,
GstPad * pad, gpointer user_data)
{
GstHarness *h;
GstHarness **sink_harness = user_data;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
h = gst_harness_new_with_element (element, NULL, GST_OBJECT_NAME (pad));
t->harnesses = g_list_prepend (t->harnesses, h);
if (sink_harness) {
*sink_harness = h;
g_cond_broadcast (&t->cond);
}
}
static void
new_jitterbuffer_set_fast_start (GstElement * rtpbin,
GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
gpointer user_data)
{
g_object_set (rtpjitterbuffer, "faststart-min-packets", 1, NULL);
}
GST_START_TEST (test_codec_preferences_negotiation_srcpad)
{
struct test_webrtc *t = test_webrtc_new ();
guint media_format_count[] = { 1, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, NULL);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
VAL_SDP_INIT (answer_non_reject, _count_non_rejected_media,
GUINT_TO_POINTER (0), &count);
GstHarness *h;
GstHarness *sink_harness = NULL;
guint i;
GstElement *rtpbin2;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_harness;
t->pad_added_data = &sink_harness;
rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
fail_unless (rtpbin2 != NULL);
g_signal_connect (rtpbin2, "new-jitterbuffer",
G_CALLBACK (new_jitterbuffer_set_fast_start), NULL);
g_object_unref (rtpbin2);
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_audio_test_src_harness (h);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
fail_if (gst_element_set_state (t->webrtc1,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
for (i = 0; i < 10; i++)
gst_harness_push_from_src (h);
g_mutex_lock (&t->lock);
while (sink_harness == NULL) {
gst_harness_push_from_src (h);
g_cond_wait_until (&t->cond, &t->lock, g_get_monotonic_time () + 5000);
}
g_mutex_unlock (&t->lock);
fail_unless (sink_harness->element == t->webrtc2);
gst_harness_set_sink_caps_str (sink_harness, OPUS_RTP_CAPS (100));
test_webrtc_reset_negotiation (t);
test_validate_sdp_full (t, &offer, &answer_non_reject, 0, FALSE);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -4048,6 +4160,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_force_second_media);
tcase_add_test (tc, test_codec_preferences_caps);
tcase_add_test (tc, test_codec_preferences_negotiation_sinkpad);
tcase_add_test (tc, test_codec_preferences_negotiation_srcpad);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);