webrtcbin: connect output stream on recv transceivers

With MR 7156, transceivers and transports are created earlier,
but for sendrecv media we could get `not-linked` errors due to
transportreceivebin not being connected to rtpbin yet when incoming
data arrives.

This condition wasn't being tested in elements_webrtcbin, but could be
reproduced in the webrtcbidirectional example. This commit now also
adds a test for this, so that this doesn't regress anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>

1 files changed, 64 insertions, 0 deletions

diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
index 1dd2ddf3c2..6ba8315f87 100644
--- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
+++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
@@ -4601,6 +4601,69 @@ _pad_added_harness (struct test_webrtc *t, GstElement * element,
   }
 }
 
+GST_START_TEST (test_audio_sendrecv)
+{
+  struct test_webrtc *t = test_webrtc_new ();
+  GstHarness *h1, *h2;
+
+  t->on_negotiation_needed = NULL;
+  t->on_ice_candidate = NULL;
+  t->on_pad_added = _pad_added_fakesink;
+
+  h1 = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL);
+  add_audio_test_src_harness (h1, 0xDEADBEEF);
+  t->harnesses = g_list_prepend (t->harnesses, h1);
+
+  h2 = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
+  add_audio_test_src_harness (h2, 0xBEEFDEAD);
+  t->harnesses = g_list_prepend (t->harnesses, h2);
+
+  VAL_SDP_INIT (no_duplicate_payloads, on_sdp_media_no_duplicate_payloads,
+      NULL, NULL);
+  guint media_format_count[] = { 1 };
+  VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
+      media_format_count, &no_duplicate_payloads);
+  VAL_SDP_INIT (count, _count_num_sdp_media, GUINT_TO_POINTER (1),
+      &media_formats);
+  const gchar *expected_offer_setup[] = { "actpass", };
+  VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, &count);
+  const gchar *expected_answer_setup[] = { "active", };
+  VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
+      &count);
+  const gchar *expected_offer_direction[] = { "sendrecv", };
+  VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
+      &offer_setup);
+  const gchar *expected_answer_direction[] = { "sendrecv", };
+  VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
+      &answer_setup);
+  GstWebRTCKind expected_kind = GST_WEBRTC_KIND_AUDIO;
+
+  g_signal_connect (t->webrtc1, "on-new-transceiver",
+      G_CALLBACK (on_new_transceiver_expected_kind),
+      GUINT_TO_POINTER (expected_kind));
+  g_signal_connect (t->webrtc2, "on-new-transceiver",
+      G_CALLBACK (on_new_transceiver_expected_kind),
+      GUINT_TO_POINTER (expected_kind));
+
+  test_validate_sdp (t, &offer, &answer);
+
+  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);
+
+  /* Exchange a few buffers between webrtcbin1 and webrtcbin2 to check
+     that they can handle incoming data and we get no errors on the bus. */
+  for (int i = 0; i < 5; i++) {
+    gst_harness_push_from_src (h1);
+    gst_harness_push_from_src (h2);
+  }
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
 static void
 new_jitterbuffer_set_fast_start (GstElement * rtpbin,
     GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@@ -6001,6 +6064,7 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_session_stats);
     tcase_add_test (tc, test_stats_with_stream);
     tcase_add_test (tc, test_audio);
+    tcase_add_test (tc, test_audio_sendrecv);
     tcase_add_test (tc, test_ice_port_restriction);
     tcase_add_test (tc, test_audio_video);
     tcase_add_test (tc, test_media_direction);