Commit 7bf18ad2 authored by Matthew Waters's avatar Matthew Waters
Browse files

webrtc: start in the closed state

This means that we will reject all operations before we've transitioned
into READY.

This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread.  Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
parent c4fe5239
......@@ -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;
}
......@@ -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);
......
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