webrtc-recv.c

static gboolean
start_pipeline (void)
{
  GstStateChangeReturn ret;
  GError *error = NULL;
  // GstCaps *audio_caps;
  // GstWebRTCRTPTransceiver *trans = NULL;
  // GArray *transceivers = NULL;

  pipe1 =
      gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
      STUN_SERVER
      TURN_SERVER
      "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
      "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
      "audiotestsrc is-live=true wave=ticks ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
      "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);

  if (error) {
    g_printerr ("Failed to parse launch: %s\n", error->message);
    g_error_free (error);
    goto err;
  }

  webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
  g_assert_nonnull (webrtc1);

  /* This is the gstwebrtc entry point where we create the offer and so on. It
   * will be called when the pipeline goes to PLAYING. */
  g_signal_connect (webrtc1, "on-negotiation-needed",
      G_CALLBACK (on_negotiation_needed), NULL);
  /* We need to transmit this ICE candidate to the browser via the websockets
   * signalling server. Incoming ice candidates from the browser need to be
   * added by us too, see on_server_message() */
  g_signal_connect (webrtc1, "on-ice-candidate",
      G_CALLBACK (send_ice_candidate_message), NULL);
  g_signal_connect (webrtc1, "notify::ice-gathering-state",
      G_CALLBACK (on_ice_gathering_state_notify), NULL);

  gst_element_set_state (pipe1, GST_STATE_READY);

  /* Incoming streams will be exposed via this signal */
  g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream), pipe1);

  // Create a 2nd transceiver for the receive only audio stream
  // audio_caps = gst_caps_from_string ("application/x-rtp,media=audio,encoding-name=OPUS,payload=97");
  // g_signal_emit_by_name (webrtc1, "add-transceiver", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, audio_caps, &trans);
  // gst_caps_unref (audio_caps);
  // if (trans != NULL) {
  //   trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
  //   gst_object_unref(trans);
  // } else {
  //   g_print ("audio transceiver null!\n");
  // }

  // g_signal_emit_by_name(webrtc1, "get-transceivers", &transceivers);
  // g_assert(transceivers != NULL && transceivers->len == 2);

  /* Lifetime is the same as the pipeline itself */
  gst_object_unref (webrtc1);

  g_print ("Starting pipeline\n");
  ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
  if (ret == GST_STATE_CHANGE_FAILURE)
    goto err;

  return TRUE;

err:
  if (pipe1)
    g_clear_object (&pipe1);
  if (webrtc1)
    webrtc1 = NULL;
  return FALSE;
}