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webrtcbin: silence spurious warning when creating answer transceiver
When creating a transceiver when creating an answer, the media kind of the transceiver was never set correctly initially. This would lead to a GST_WARNING being produced about changin a transceiver's media kind. Fix by retrieving the GstSDPMedia kind from the offer instead as the answer GstSDPMedia has not been set as the answer caps have not been chosen yet. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
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1 changed files with 18 additions and 0 deletions
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@ -1022,6 +1022,16 @@ create_audio_test (void)
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return t;
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return t;
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}
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}
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static void
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on_new_transceiver_expected_kind (GstWebRTCBin * webrtc,
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GstWebRTCRTPTransceiver * trans, gpointer user_data)
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{
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GstWebRTCKind kind, expected = GPOINTER_TO_UINT (user_data);
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g_object_get (trans, "kind", &kind, NULL);
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fail_unless_equals_int (kind, expected);
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}
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GST_START_TEST (test_audio)
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GST_START_TEST (test_audio)
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{
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{
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struct test_webrtc *t = create_audio_test ();
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struct test_webrtc *t = create_audio_test ();
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@ -1043,10 +1053,18 @@ GST_START_TEST (test_audio)
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const gchar *expected_answer_direction[] = { "recvonly", };
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const gchar *expected_answer_direction[] = { "recvonly", };
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VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
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VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
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&answer_setup);
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&answer_setup);
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GstWebRTCKind expected_kind = GST_WEBRTC_KIND_AUDIO;
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/* check that a single stream connection creates the associated number
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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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* of media sections */
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g_signal_connect (t->webrtc1, "on-new-transceiver",
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G_CALLBACK (on_new_transceiver_expected_kind),
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GUINT_TO_POINTER (expected_kind));
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g_signal_connect (t->webrtc2, "on-new-transceiver",
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G_CALLBACK (on_new_transceiver_expected_kind),
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GUINT_TO_POINTER (expected_kind));
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test_validate_sdp (t, &offer, &answer);
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test_validate_sdp (t, &offer, &answer);
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test_webrtc_free (t);
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test_webrtc_free (t);
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}
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}
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