From 2aa7efedd3ea3ab73eb7884ae569098418065e55 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Wed, 5 May 2021 19:18:02 -0400 Subject: webrtc test: Add test for codec preferences negotiation Validate that it does the intersection with the caps from the sink pad and rejects the offer creation otherwise. Part-of: --- tests/check/elements/webrtcbin.c | 79 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 77 insertions(+), 2 deletions(-) (limited to 'tests/check') diff --git a/tests/check/elements/webrtcbin.c b/tests/check/elements/webrtcbin.c index fba357239..2f6747f20 100644 --- a/tests/check/elements/webrtcbin.c +++ b/tests/check/elements/webrtcbin.c @@ -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); -- cgit v1.2.1