mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-20 13:06:23 +00:00
tests/webrtc: factor out sdp validation into a single function
This commit is contained in:
Matthew Waters
parent
eb79f95bf8
commit
ebb9c3c298
1 changed files with 61 additions and 235 deletions
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@ -664,6 +664,36 @@ _check_validate_sdp (struct test_webrtc *t, GstElement * element,
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return offer;
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}
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static void
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test_validate_sdp (struct test_webrtc *t, struct validate_sdp *offer,
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struct validate_sdp *answer)
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{
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if (offer) {
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t->offer_data = offer;
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t->on_offer_created = _check_validate_sdp;
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} else {
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t->offer_data = NULL;
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t->on_offer_created = NULL;
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}
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if (answer) {
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t->answer_data = answer;
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t->on_answer_created = _check_validate_sdp;
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} else {
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t->answer_data = NULL;
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t->on_answer_created = NULL;
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}
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless (t->state == STATE_ANSWER_CREATED);
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}
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GST_START_TEST (test_sdp_no_media)
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{
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struct test_webrtc *t = test_webrtc_new ();
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@ -675,20 +705,8 @@ GST_START_TEST (test_sdp_no_media)
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/* check that a no stream connection creates 0 media sections */
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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test_validate_sdp (t, &offer, &answer);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless (t->state == STATE_ANSWER_CREATED);
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test_webrtc_free (t);
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}
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@ -721,6 +739,7 @@ create_audio_test (void)
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GstHarness *h;
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t->on_negotiation_needed = NULL;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_fakesink;
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h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
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@ -741,22 +760,7 @@ GST_START_TEST (test_audio)
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/* check that a single stream connection creates the associated number
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* of media sections */
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
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}
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@ -765,16 +769,9 @@ GST_END_TEST;
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static struct test_webrtc *
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create_audio_video_test (void)
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{
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struct test_webrtc *t = test_webrtc_new ();
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struct test_webrtc *t = create_audio_test ();
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GstHarness *h;
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t->on_negotiation_needed = NULL;
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t->on_pad_added = _pad_added_fakesink;
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h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
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add_fake_audio_src_harness (h, 96);
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t->harnesses = g_list_prepend (t->harnesses, h);
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h = gst_harness_new_with_element (t->webrtc1, "sink_1", NULL);
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add_fake_video_src_harness (h, 97);
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t->harnesses = g_list_prepend (t->harnesses, h);
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@ -793,22 +790,7 @@ GST_START_TEST (test_audio_video)
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/* check that a dual stream connection creates the associated number
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* of media sections */
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
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}
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@ -860,9 +842,14 @@ GST_START_TEST (test_media_direction)
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struct test_webrtc *t = create_audio_video_test ();
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const gchar *expected_offer[] = { "sendrecv", "sendrecv" };
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const gchar *expected_answer[] = { "sendrecv", "recvonly" };
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struct validate_sdp offer = { on_sdp_media_direction, expected_offer, NULL };
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struct validate_sdp answer =
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struct validate_sdp offer_direction =
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{ on_sdp_media_direction, expected_offer, NULL };
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struct validate_sdp offer =
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{ _count_num_sdp_media, GUINT_TO_POINTER (2), &offer_direction };
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struct validate_sdp answer_direction =
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{ on_sdp_media_direction, expected_answer, NULL };
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struct validate_sdp answer =
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{ _count_num_sdp_media, GUINT_TO_POINTER (2), &answer_direction };
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GstHarness *h;
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/* check the default media directions for transceivers */
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@ -871,22 +858,7 @@ GST_START_TEST (test_media_direction)
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add_fake_audio_src_harness (h, 96);
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t->harnesses = g_list_prepend (t->harnesses, h);
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
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}
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@ -899,8 +871,6 @@ on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
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const GstSDPMedia *vmedia;
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guint j;
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fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), 2);
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vmedia = gst_sdp_message_get_media (desc->sdp, 1);
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for (j = 0; j < gst_sdp_media_attributes_len (vmedia); j++) {
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@ -927,17 +897,12 @@ on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
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GST_START_TEST (test_payload_types)
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{
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struct test_webrtc *t = create_audio_video_test ();
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struct validate_sdp offer = { on_sdp_media_payload_types, NULL, NULL };
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struct validate_sdp payloads = { on_sdp_media_payload_types, NULL, NULL };
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struct validate_sdp offer =
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{ _count_num_sdp_media, GUINT_TO_POINTER (2), &payloads };
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GstWebRTCRTPTransceiver *trans;
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GArray *transceivers;
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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/* We don't really care about the answer here */
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t->on_answer_created = NULL;
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g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
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fail_unless_equals_int (transceivers->len, 2);
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trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
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@ -945,15 +910,8 @@ GST_START_TEST (test_payload_types)
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NULL);
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g_array_unref (transceivers);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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/* We don't really care about the answer here */
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test_validate_sdp (t, &offer, NULL);
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test_webrtc_free (t);
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}
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@ -994,23 +952,7 @@ GST_START_TEST (test_media_setup)
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struct validate_sdp answer = { on_sdp_media_setup, expected_answer, NULL };
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/* check the default dtls setup negotiation values */
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
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}
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@ -1322,20 +1264,8 @@ GST_START_TEST (test_session_stats)
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GstPromise *p;
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/* test that the stats generated without any streams are sane */
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t->on_negotiation_needed = NULL;
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t->on_offer_created = NULL;
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t->on_answer_created = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, NULL, NULL);
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p = gst_promise_new_with_change_func (_on_stats, t, NULL);
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g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
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@ -1403,15 +1333,9 @@ GST_START_TEST (test_add_recvonly_transceiver)
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/* add a transceiver that will only receive an opus stream and check that
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* the created offer is marked as recvonly */
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t->on_negotiation_needed = NULL;
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t->on_pad_added = _pad_added_fakesink;
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_fakesink;
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/* setup recvonly transceiver */
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caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
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@ -1426,16 +1350,8 @@ GST_START_TEST (test_add_recvonly_transceiver)
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h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
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add_fake_audio_src_harness (h, 96);
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t->harnesses = g_list_prepend (t->harnesses, h);
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test_validate_sdp (t, &offer, &answer);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_webrtc_free (t);
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}
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@ -1457,15 +1373,9 @@ GST_START_TEST (test_recvonly_sendonly)
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/* add a transceiver that will only receive an opus stream and check that
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* the created offer is marked as recvonly */
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t->on_negotiation_needed = NULL;
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t->on_pad_added = _pad_added_fakesink;
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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t->on_pad_added = _pad_added_fakesink;
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/* setup recvonly transceiver */
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caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
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@ -1492,15 +1402,8 @@ GST_START_TEST (test_recvonly_sendonly)
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add_fake_audio_src_harness (h, 96);
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t->harnesses = g_list_prepend (t->harnesses, h);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_webrtc_free (t);
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}
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@ -1540,10 +1443,6 @@ GST_START_TEST (test_data_channel_create)
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gchar *label;
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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@ -1559,10 +1458,8 @@ GST_START_TEST (test_data_channel_create)
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g_signal_connect (channel, "on-error",
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G_CALLBACK (on_channel_error_not_reached), NULL);
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test_webrtc_create_offer (t, t->webrtc1);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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g_object_unref (channel);
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g_free (label);
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test_webrtc_free (t);
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@ -2145,22 +2042,7 @@ GST_START_TEST (test_bundle_audio_video_max_bundle_max_bundle)
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gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
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"max-bundle");
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t->on_negotiation_needed = NULL;
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t->offer_data = &offer;
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t->on_offer_created = _check_validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = _check_validate_sdp;
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t->on_ice_candidate = NULL;
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
|
||||
}
|
||||
|
|
@ -2202,22 +2084,7 @@ GST_START_TEST (test_bundle_audio_video_max_compat_max_bundle)
|
|||
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
|
||||
"max-bundle");
|
||||
|
||||
t->on_negotiation_needed = NULL;
|
||||
t->offer_data = &offer;
|
||||
t->on_offer_created = _check_validate_sdp;
|
||||
t->answer_data = &answer;
|
||||
t->on_answer_created = _check_validate_sdp;
|
||||
t->on_ice_candidate = NULL;
|
||||
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
|
||||
test_webrtc_create_offer (t, t->webrtc1);
|
||||
|
||||
test_webrtc_wait_for_answer_error_eos (t);
|
||||
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
|
||||
test_validate_sdp (t, &offer, &answer);
|
||||
|
||||
test_webrtc_free (t);
|
||||
}
|
||||
|
|
@ -2260,22 +2127,7 @@ GST_START_TEST (test_bundle_audio_video_max_bundle_none)
|
|||
"max-bundle");
|
||||
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy", "none");
|
||||
|
||||
t->on_negotiation_needed = NULL;
|
||||
t->offer_data = &offer;
|
||||
t->on_offer_created = _check_validate_sdp;
|
||||
t->answer_data = &answer;
|
||||
t->on_answer_created = _check_validate_sdp;
|
||||
t->on_ice_candidate = NULL;
|
||||
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
|
||||
test_webrtc_create_offer (t, t->webrtc1);
|
||||
|
||||
test_webrtc_wait_for_answer_error_eos (t);
|
||||
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
|
||||
test_validate_sdp (t, &offer, &answer);
|
||||
|
||||
test_webrtc_free (t);
|
||||
}
|
||||
|
|
@ -2319,13 +2171,6 @@ GST_START_TEST (test_bundle_audio_video_data)
|
|||
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
|
||||
"max-bundle");
|
||||
|
||||
t->on_negotiation_needed = NULL;
|
||||
t->offer_data = &offer;
|
||||
t->on_offer_created = _check_validate_sdp;
|
||||
t->answer_data = &answer;
|
||||
t->on_answer_created = _check_validate_sdp;
|
||||
t->on_ice_candidate = NULL;
|
||||
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
|
|
@ -2334,10 +2179,7 @@ GST_START_TEST (test_bundle_audio_video_data)
|
|||
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
|
||||
&channel);
|
||||
|
||||
test_webrtc_create_offer (t, t->webrtc1);
|
||||
|
||||
test_webrtc_wait_for_answer_error_eos (t);
|
||||
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
|
||||
test_validate_sdp (t, &offer, &answer);
|
||||
|
||||
g_object_unref (channel);
|
||||
test_webrtc_free (t);
|
||||
|
|
@ -2362,26 +2204,10 @@ GST_START_TEST (test_duplicate_nego)
|
|||
t->harnesses = g_list_prepend (t->harnesses, h);
|
||||
|
||||
t->on_negotiation_needed = NULL;
|
||||
t->offer_data = &offer;
|
||||
t->on_offer_created = _check_validate_sdp;
|
||||
t->answer_data = &answer;
|
||||
t->on_answer_created = _check_validate_sdp;
|
||||
t->on_ice_candidate = NULL;
|
||||
|
||||
fail_if (gst_element_set_state (t->webrtc1,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
fail_if (gst_element_set_state (t->webrtc2,
|
||||
GST_STATE_READY) == GST_STATE_CHANGE_FAILURE);
|
||||
|
||||
test_webrtc_create_offer (t, t->webrtc1);
|
||||
test_webrtc_wait_for_answer_error_eos (t);
|
||||
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
|
||||
|
||||
test_validate_sdp (t, &offer, &answer);
|
||||
test_webrtc_signal_state (t, STATE_NEGOTIATION_NEEDED);
|
||||
test_validate_sdp (t, &offer, &answer);
|
||||
|
||||
test_webrtc_create_offer (t, t->webrtc1);
|
||||
test_webrtc_wait_for_answer_error_eos (t);
|
||||
fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
|
||||
test_webrtc_free (t);
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue