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webrtc tests: Verify that create-offer is rejected when needed

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Verify that it gets rejected if a m-line at index 1 is requested but
there is no m-line 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-30 16:04:33 -04:00
parent 913d308e22
commit 8df5b9f974
1 changed files with 80 additions and 0 deletions
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80
tests/check/elements/webrtcbin.c
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@ -3456,6 +3456,85 @@ GST_START_TEST (test_reject_request_pad)
GST_END_TEST;
static void
_verify_media_types (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
gchar **media_types = user_data;
int i;
for (i = 0; i < gst_sdp_message_medias_len (desc->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (desc->sdp, i);
fail_unless_equals_string (gst_sdp_media_get_media (media), media_types[i]);
}
}
GST_START_TEST (test_reject_create_offer)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
const gchar *media_types[] = { "video", "audio" };
VAL_SDP_INIT (media_type, _verify_media_types, &media_types, NULL);
guint media_format_count[] = { 1, 1 };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &media_type);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2),
&media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &count);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&payloads);
const gchar *expected_answer_setup[] = { "active", "active" };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&payloads);
const gchar *expected_offer_direction[] = { "sendrecv", "sendrecv" };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "recvonly", "recvonly" };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup sendonly peer */
h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Check that if there is no 0, we can't create an offer with a hole */
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc1, "create-offer", NULL, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
fail_unless (s != NULL);
fail_unless (gst_structure_has_name (s, "application/x-gstwebrtcbin-error"));
gst_structure_get (s, "error", G_TYPE_ERROR, &error, NULL);
fail_unless (g_error_matches (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION));
g_clear_error (&error);
gst_promise_unref (promise);
h = gst_harness_new_with_element (t->webrtc1, "sink_%u", NULL);
add_fake_video_src_harness (h, 97);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Adding a second sink, which will fill m-line 0, should fix it */
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -3500,6 +3579,7 @@ webrtcbin_suite (void)
tcase_add_test (tc,
test_bundle_codec_preferences_rtx_no_duplicate_payloads);
tcase_add_test (tc, test_reject_request_pad);
tcase_add_test (tc, test_reject_create_offer);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);