mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
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.
This commit is contained in:
Matthew Waters
parent
c4fe52395b
commit
7bf18ad258
2 changed files with 116 additions and 12 deletions
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@ -5146,4 +5146,7 @@ gst_webrtc_bin_init (GstWebRTCBin * webrtc)
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g_array_new (FALSE, TRUE, sizeof (IceCandidateItem *));
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g_array_set_clear_func (webrtc->priv->pending_ice_candidates,
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(GDestroyNotify) _clear_ice_candidate_item);
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/* we start off closed until we move to READY */
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webrtc->priv->is_closed = TRUE;
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}
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@ -643,11 +643,17 @@ 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 = GUINT_TO_POINTER (0);
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t->on_offer_created = _count_num_sdp_media;
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t->answer_data = GUINT_TO_POINTER (0);
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t->on_answer_created = _count_num_sdp_media;
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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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@ -700,12 +706,18 @@ 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 = GUINT_TO_POINTER (1);
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t->on_offer_created = _count_num_sdp_media;
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t->answer_data = GUINT_TO_POINTER (1);
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t->on_answer_created = _count_num_sdp_media;
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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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@ -742,12 +754,18 @@ 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 = GUINT_TO_POINTER (2);
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t->on_offer_created = _count_num_sdp_media;
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t->answer_data = GUINT_TO_POINTER (2);
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t->on_answer_created = _count_num_sdp_media;
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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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@ -842,12 +860,18 @@ 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 = validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = 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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@ -896,6 +920,7 @@ GST_START_TEST (test_payload_types)
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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 = validate_sdp;
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t->on_ice_candidate = NULL;
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@ -909,6 +934,11 @@ 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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@ -954,12 +984,18 @@ GST_START_TEST (test_media_setup)
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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 = validate_sdp;
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t->answer_data = &answer;
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t->on_answer_created = 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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@ -1276,9 +1312,15 @@ GST_START_TEST (test_session_stats)
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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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@ -1350,6 +1392,7 @@ 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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@ -1372,6 +1415,11 @@ GST_START_TEST (test_add_recvonly_transceiver)
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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_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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@ -1397,6 +1445,7 @@ 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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@ -1430,6 +1479,11 @@ 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_webrtc_create_offer (t, t->webrtc1);
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test_webrtc_wait_for_answer_error_eos (t);
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@ -1505,6 +1559,11 @@ GST_START_TEST (test_data_channel_create)
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t->on_answer_created = 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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1560,6 +1619,11 @@ GST_START_TEST (test_data_channel_remote_notify)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1567,8 +1631,10 @@ GST_START_TEST (test_data_channel_remote_notify)
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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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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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@ -1628,6 +1694,11 @@ GST_START_TEST (test_data_channel_transfer_string)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel_transfer_string;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1635,8 +1706,10 @@ GST_START_TEST (test_data_channel_transfer_string)
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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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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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@ -1703,6 +1776,11 @@ GST_START_TEST (test_data_channel_transfer_data)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel_transfer_data;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1710,8 +1788,10 @@ GST_START_TEST (test_data_channel_transfer_data)
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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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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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@ -1754,6 +1834,11 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel_create_data_channel;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "prev-label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1761,8 +1846,10 @@ GST_START_TEST (test_data_channel_create_after_negotiate)
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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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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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@ -1808,6 +1895,11 @@ GST_START_TEST (test_data_channel_low_threshold)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel_check_low_threshold_emitted;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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@ -1815,8 +1907,10 @@ GST_START_TEST (test_data_channel_low_threshold)
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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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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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@ -1874,13 +1968,20 @@ GST_START_TEST (test_data_channel_max_message_size)
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t->on_ice_candidate = NULL;
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t->on_data_channel = have_data_channel_transfer_large_data;
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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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g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
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&channel);
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g_assert_nonnull (channel);
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t->data_channel_data = channel;
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gst_element_set_state (t->webrtc1, GST_STATE_PLAYING);
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gst_element_set_state (t->webrtc2, GST_STATE_PLAYING);
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fail_if (gst_element_set_state (t->webrtc1,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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fail_if (gst_element_set_state (t->webrtc2,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
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test_webrtc_create_offer (t, t->webrtc1);
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