webrtc-sendrecv.c

static void
handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
    const char *sink_name)
{
  GstPad *qpad, *rtp_sink, *rtp_src, *rtcp_src, *rtp_udp_sink, *rtcp_udp_sink;
  GstElement *q, *conv, *resample, *opusenc, *rtpopuspay, *rtpbin, *rtp_udpsink, *rtcp_udpsink;
  GstPadLinkReturn ret;
  gchar *rtpopuspay_string, *rtp_udpsink_string, *rtcp_udpsink_string;

  g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
  if (g_strcmp0 (convert_name, "audioconvert") != 0) {
    g_printerr ("peer video input not supported");
    return;
  }

  // data is queued until limits specified by the 
  // max-size-buffers, max-size-bytes and/or max-size-time properties has been reached. 
  // Any attempt to push more buffers into the queue will block 
  // the pushing thread until more space becomes available.
  q = gst_element_factory_make ("queue", NULL);
  g_assert_nonnull (q);

  // converts raw audio buffers between various possible formats.
  // It supports integer to float conversion, width/depth conversion,
  // signedness and endianness conversion and 
  // channel transformations (ie. upmixing and downmixing), 
  // as well as dithering and noise-shaping.
  conv = gst_element_factory_make (convert_name, NULL);
  g_assert_nonnull (conv);

  /* Might also need to resample, so add it just in case.
    * Will be a no-op if it's not required. */
  // TODO: resample to 16,000 hz ?
  resample = gst_element_factory_make ("audioresample", NULL);
  g_assert_nonnull (resample);

  // encode to opus
  opusenc = gst_element_factory_make ("opusenc inband-fec=true bitrate-type=vbr", NULL);
  g_assert_nonnull (opusenc);

  // Puts Opus audio in RTP packets
  rtpopuspay_string = g_strdup_printf ("rtpopuspay pt=%d ssrc=%d", audio_pt, audio_ssrc);
  rtpopuspay = gst_element_factory_make (rtpopuspay_string, NULL);
  g_assert_nonnull (rtpopuspay);
  g_free (rtpopuspay_string);

  // bin for forwarding RTP packets
  rtpbin = gst_element_factory_make ("rtpbin", NULL);
  g_assert_nonnull (rtpbin);

  gst_bin_add_many (GST_BIN (pipe), q, conv, resample, opusenc, rtpopuspay, rtpbin, NULL);
  gst_element_sync_state_with_parent (q);
  gst_element_sync_state_with_parent (conv);
  gst_element_sync_state_with_parent (resample);
  gst_element_sync_state_with_parent (opusenc);
  gst_element_sync_state_with_parent (rtpopuspay);
  gst_element_sync_state_with_parent (rtpbin);

  rtp_sink = gst_element_get_request_pad (rtpbin, "send_rtp_sink_1");
  gst_element_link_many (q, conv, resample, opusenc, rtpopuspay, rtp_sink, NULL);

  // create pad for beginning of chain created above
  qpad = gst_element_get_static_pad (q, "sink");
  ret = gst_pad_link (pad, qpad);
  g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);

  // create udpsinks for RTP and RTCP
  rtp_udpsink_string = g_strdup_printf ("udpsink host=%s port=%s", audio_transport_ip, audio_transport_rtp_port);
  rtp_udpsink = gst_element_factory_make (rtp_udpsink_string, NULL);
  g_assert_nonnull (rtp_udpsink);
  g_free (rtp_udpsink_string);
  rtcp_udpsink_string = g_strdup_printf ("udpsink host=%s port=%s sync=false async=false", audio_transport_ip, audio_transport_rtcp_port);
  rtcp_udpsink = gst_element_factory_make (rtcp_udpsink_string, NULL);
  g_assert_nonnull (rtcp_udpsink);
  g_free (rtcp_udpsink_string);

  gst_bin_add_many (GST_BIN (pipe), rtp_udpsink, rtcp_udpsink, NULL);
  gst_element_sync_state_with_parent (rtp_udpsink);
  gst_element_sync_state_with_parent (rtcp_udpsink);

  // send_rtp_src_1 created when send_rtp_sink_1 was requested
  rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_1");

  // need to request send_rtcp_src_1
  rtcp_src = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1");

  // link the RTP and RTCP pads
  rtp_udp_sink = gst_element_get_static_pad (rtp_udpsink, "sink");
  rtcp_udp_sink = gst_element_get_static_pad (rtcp_udpsink, "sink");
  ret = gst_pad_link (rtp_src, rtp_udp_sink);
  g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);
  ret = gst_pad_link (rtcp_src, rtcp_udp_sink);
  g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);
}