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webrtc test: Add test for codec preferences negotiation

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Validate that it does the intersection with the caps from
the sink pad and rejects the offer creation otherwise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
This commit is contained in:
Olivier Crête 2021-05-05 19:18:02 -04:00
parent f6345b4b03
commit 2aa7efedd3
1 changed files with 77 additions and 2 deletions
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79
tests/check/elements/webrtcbin.c
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@ -1188,9 +1188,10 @@ on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * desc, gpointer user_data)
{
const GstSDPMedia *vmedia;
guint video_mline = GPOINTER_TO_UINT (user_data);
guint j;
vmedia = gst_sdp_message_get_media (desc->sdp, 1);
vmedia = gst_sdp_message_get_media (desc->sdp, video_mline);
for (j = 0; j < gst_sdp_media_attributes_len (vmedia); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (vmedia, j);
@ -1221,7 +1222,8 @@ GST_START_TEST (test_payload_types)
guint media_format_count[] = { 1, 5, };
VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
media_format_count, &no_duplicate_payloads);
VAL_SDP_INIT (payloads, on_sdp_media_payload_types, NULL, &media_formats);
VAL_SDP_INIT (payloads, on_sdp_media_payload_types, GUINT_TO_POINTER (1),
&media_formats);
VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads);
const gchar *expected_offer_setup[] = { "actpass", "actpass" };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
@ -3917,6 +3919,78 @@ GST_START_TEST (test_codec_preferences_caps)
GST_END_TEST;
GST_START_TEST (test_codec_preferences_negotiation_sinkpad)
{
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 (payloads2, on_sdp_media_payload_types, GUINT_TO_POINTER (0),
&count);
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, &payloads2);
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);
GstPad *pad;
GstWebRTCRTPTransceiver *transceiver;
GstHarness *h;
GstCaps *caps;
GstPromise *promise;
GstPromiseResult res;
const GstStructure *s;
GError *error = NULL;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
g_object_get (pad, "transceiver", &transceiver, NULL);
caps = gst_caps_from_string (VP8_RTP_CAPS (115) ";" VP8_RTP_CAPS (97));
g_object_set (transceiver, "codec-preferences", caps, NULL);
gst_caps_unref (caps);
gst_object_unref (transceiver);
gst_object_unref (pad);
add_fake_video_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
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_CAPS_NEGOTIATION_FAILED));
g_clear_error (&error);
gst_promise_unref (promise);
caps = gst_caps_from_string (VP8_RTP_CAPS (97));
gst_harness_set_src_caps (h, caps);
test_validate_sdp (t, &offer, &answer);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -3965,6 +4039,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_reject_set_description);
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);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);