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webrtc test: Verify that forcing different kinds on peers fails

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If the offer contains an audio kind and a video kind, forcing them both
at m-line zero will fail.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-30 16:16:50 -04:00
parent 8df5b9f974
commit 2bd647e999
1 changed files with 69 additions and 0 deletions
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69
tests/check/elements/webrtcbin.c
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@ -3535,6 +3535,74 @@ GST_START_TEST (test_reject_create_offer)
GST_END_TEST;
GST_START_TEST (test_reject_set_description)
{
struct test_webrtc *t = test_webrtc_new ();
GstHarness *h;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
GstWebRTCSessionDescription *desc = NULL;
GstPadTemplate *templ;
GstCaps *caps;
GstPad *pad;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
/* setup peer 1 */
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
/* Create a second side with specific video caps */
templ = gst_element_get_pad_template (t->webrtc2, "sink_%u");
fail_unless (templ != NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
pad = gst_element_request_pad (t->webrtc2, templ, "sink_0", caps);
fail_unless (pad != NULL);
gst_caps_unref (caps);
gst_object_unref (pad);
/* Create an offer */
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-gst-promise"));
gst_structure_get (s, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &desc,
NULL);
fail_unless (desc != NULL);
gst_promise_unref (promise);
fail_if (gst_element_set_state (t->webrtc2,
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
/* Verify that setting an offer where there is a forced m-line with
a different kind fails. */
promise = gst_promise_new ();
g_signal_emit_by_name (t->webrtc2, "set-remote-description", desc, promise);
res = gst_promise_wait (promise);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
s = gst_promise_get_reply (promise);
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);
fail_unless (s != NULL);
gst_promise_unref (promise);
gst_webrtc_session_description_free (desc);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -3580,6 +3648,7 @@ webrtcbin_suite (void)
test_bundle_codec_preferences_rtx_no_duplicate_payloads);
tcase_add_test (tc, test_reject_request_pad);
tcase_add_test (tc, test_reject_create_offer);
tcase_add_test (tc, test_reject_set_description);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);