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-rw-r--r--ext/webrtc/gstwebrtcbin.c3
-rw-r--r--tests/check/elements/webrtcbin.c125
2 files changed, 116 insertions, 12 deletions
diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index d7c1d479b..0bb1eb127 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -5146,4 +5146,7 @@ gst_webrtc_bin_init (GstWebRTCBin * webrtc)
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem *));
g_array_set_clear_func (webrtc->priv->pending_ice_candidates,
(GDestroyNotify) _clear_ice_candidate_item);
+
+ /* we start off closed until we move to READY */
+ webrtc->priv->is_closed = TRUE;
}
diff --git a/tests/check/elements/webrtcbin.c b/tests/check/elements/webrtcbin.c
index e6316687a..2f3420001 100644
--- a/tests/check/elements/webrtcbin.c
+++ b/tests/check/elements/webrtcbin.c
@@ -643,11 +643,17 @@ GST_START_TEST (test_sdp_no_media)
/* check that a no stream connection creates 0 media sections */
+ t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (0);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (0);
t->on_answer_created = _count_num_sdp_media;
+ 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);
@@ -700,12 +706,18 @@ GST_START_TEST (test_audio)
/* check that a single stream connection creates the associated number
* of media sections */
+ t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (1);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (1);
t->on_answer_created = _count_num_sdp_media;
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);
@@ -742,12 +754,18 @@ GST_START_TEST (test_audio_video)
/* check that a dual stream connection creates the associated number
* of media sections */
+ t->on_negotiation_needed = NULL;
t->offer_data = GUINT_TO_POINTER (2);
t->on_offer_created = _count_num_sdp_media;
t->answer_data = GUINT_TO_POINTER (2);
t->on_answer_created = _count_num_sdp_media;
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);
@@ -842,12 +860,18 @@ GST_START_TEST (test_media_direction)
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
+ t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = validate_sdp;
t->answer_data = &answer;
t->on_answer_created = 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);
@@ -896,6 +920,7 @@ GST_START_TEST (test_payload_types)
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
+ t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = validate_sdp;
t->on_ice_candidate = NULL;
@@ -909,6 +934,11 @@ GST_START_TEST (test_payload_types)
NULL);
g_array_unref (transceivers);
+ 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);
@@ -954,12 +984,18 @@ GST_START_TEST (test_media_setup)
/* check the default dtls setup negotiation values */
+ t->on_negotiation_needed = NULL;
t->offer_data = &offer;
t->on_offer_created = validate_sdp;
t->answer_data = &answer;
t->on_answer_created = 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);
@@ -1276,9 +1312,15 @@ GST_START_TEST (test_session_stats)
/* test that the stats generated without any streams are sane */
+ t->on_negotiation_needed = NULL;
t->on_offer_created = NULL;
t->on_answer_created = 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);
@@ -1350,6 +1392,7 @@ GST_START_TEST (test_add_recvonly_transceiver)
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
+ t->on_negotiation_needed = NULL;
t->on_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
@@ -1372,6 +1415,11 @@ GST_START_TEST (test_add_recvonly_transceiver)
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
+ 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);
@@ -1397,6 +1445,7 @@ GST_START_TEST (test_recvonly_sendonly)
/* add a transceiver that will only receive an opus stream and check that
* the created offer is marked as recvonly */
+ t->on_negotiation_needed = NULL;
t->on_pad_added = _pad_added_fakesink;
t->on_negotiation_needed = NULL;
t->offer_data = &offer;
@@ -1430,6 +1479,11 @@ GST_START_TEST (test_recvonly_sendonly)
add_fake_audio_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
+ 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);
@@ -1505,6 +1559,11 @@ GST_START_TEST (test_data_channel_create)
t->on_answer_created = 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1560,6 +1619,11 @@ GST_START_TEST (test_data_channel_remote_notify)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1567,8 +1631,10 @@ GST_START_TEST (test_data_channel_remote_notify)
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
@@ -1628,6 +1694,11 @@ GST_START_TEST (test_data_channel_transfer_string)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_string;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1635,8 +1706,10 @@ GST_START_TEST (test_data_channel_transfer_string)
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
@@ -1703,6 +1776,11 @@ GST_START_TEST (test_data_channel_transfer_data)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_data;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1710,8 +1788,10 @@ GST_START_TEST (test_data_channel_transfer_data)
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
@@ -1754,6 +1834,11 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_create_data_channel;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "prev-label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1761,8 +1846,10 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
@@ -1808,6 +1895,11 @@ GST_START_TEST (test_data_channel_low_threshold)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_check_low_threshold_emitted;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
@@ -1815,8 +1907,10 @@ GST_START_TEST (test_data_channel_low_threshold)
g_signal_connect (channel, "on-error",
G_CALLBACK (on_channel_error_not_reached), NULL);
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
@@ -1874,13 +1968,20 @@ GST_START_TEST (test_data_channel_max_message_size)
t->on_ice_candidate = NULL;
t->on_data_channel = have_data_channel_transfer_large_data;
+ 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);
+
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
t->data_channel_data = channel;
- gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
- gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
+ fail_if (gst_element_set_state (t->webrtc1,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
+ fail_if (gst_element_set_state (t->webrtc2,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
test_webrtc_create_offer (t, t->webrtc1);
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-23 08:05:13 +0000