C++ (Cpp) gst_srtp_get_soft_limit_reached Examples

Example #1

File: gstsrtpdec.c Project: fanc999/gst-plugins-bad

static GstFlowReturn
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
    gboolean is_rtcp)
{
  GstSrtpDec *filter = GST_SRTP_DEC (parent);
  GstPad *otherpad;
  GstSrtpDecSsrcStream *stream = NULL;
  GstFlowReturn ret = GST_FLOW_OK;
  guint32 ssrc = 0;

  GST_OBJECT_LOCK (filter);

  /* Check if this stream exists, if not create a new stream */

  if (!(stream = validate_buffer (filter, buf, &ssrc, &is_rtcp))) {
    GST_OBJECT_UNLOCK (filter);
    GST_WARNING_OBJECT (filter, "Invalid buffer, dropping");
    goto drop_buffer;
  }

  if (!STREAM_HAS_CRYPTO (stream)) {
    GST_OBJECT_UNLOCK (filter);
    goto push_out;
  }

  if (!gst_srtp_dec_decode_buffer (filter, pad, buf, is_rtcp, ssrc)) {
    GST_OBJECT_UNLOCK (filter);
    goto drop_buffer;
  }

  GST_OBJECT_UNLOCK (filter);

  /* If all is well, we may have reached soft limit */
  if (gst_srtp_get_soft_limit_reached ())
    request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);

push_out:
  /* Push buffer to source pad */
  if (is_rtcp) {
    otherpad = filter->rtcp_srcpad;
    if (!filter->rtcp_has_segment)
      gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
          filter->rtp_srcpad, TRUE);
  } else {
    otherpad = filter->rtp_srcpad;
    if (!filter->rtp_has_segment)
      gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
          filter->rtcp_srcpad, FALSE);
  }
  ret = gst_pad_push (otherpad, buf);

  return ret;

drop_buffer:
  /* Drop buffer, except if gst_pad_push returned OK or an error */

  gst_buffer_unref (buf);

  return ret;
}

Example #2

File: gstsrtpdec.c Project: jledet/gst-plugins-bad

static GstFlowReturn
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
    gboolean is_rtcp)
{
  GstSrtpDec *filter = GST_SRTP_DEC (parent);
  GstPad *otherpad;
  err_status_t err = err_status_ok;
  GstSrtpDecSsrcStream *stream = NULL;
  GstFlowReturn ret = GST_FLOW_OK;
  gint size;
  guint32 ssrc = 0;
  GstMapInfo map;

  GST_OBJECT_LOCK (filter);

  /* Check if this stream exists, if not create a new stream */

  if (!(stream = validate_buffer (filter, buf, &ssrc, &is_rtcp))) {
    GST_OBJECT_UNLOCK (filter);
    GST_WARNING_OBJECT (filter, "Invalid buffer, dropping");
    goto drop_buffer;
  }

  if (!STREAM_HAS_CRYPTO (stream)) {
    GST_OBJECT_UNLOCK (filter);
    goto push_out;
  }

  GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT
      " with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf),
      ssrc);

  /* Change buffer to remove protection */
  buf = gst_buffer_make_writable (buf);

unprotect:

  gst_buffer_map (buf, &map, GST_MAP_READWRITE);
  size = map.size;

  gst_srtp_init_event_reporter ();

  if (is_rtcp)
    err = srtp_unprotect_rtcp (filter->session, map.data, &size);
  else {
    /* If ROC has changed, we know we need to set the initial RTP
     * sequence number too. */
    if (filter->roc_changed) {
      srtp_stream_t stream;

      stream = srtp_get_stream (filter->session, htonl (ssrc));

      if (stream) {
        guint16 seqnum = 0;
        GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;

        gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
        seqnum = gst_rtp_buffer_get_seq (&rtpbuf);
        gst_rtp_buffer_unmap (&rtpbuf);

        /* We finally add the RTP sequence number to the current
         * rollover counter. */
        stream->rtp_rdbx.index &= ~0xFFFF;
        stream->rtp_rdbx.index |= seqnum;
      }

      filter->roc_changed = FALSE;
    }
    err = srtp_unprotect (filter->session, map.data, &size);
  }

  gst_buffer_unmap (buf, &map);

  GST_OBJECT_UNLOCK (filter);

  if (err != err_status_ok) {
    GST_WARNING_OBJECT (pad,
        "Unable to unprotect buffer (unprotect failed code %d)", err);

    /* Signal user depending on type of error */
    switch (err) {
      case err_status_key_expired:
        GST_OBJECT_LOCK (filter);

        /* Update stream */
        if (find_stream_by_ssrc (filter, ssrc)) {
          GST_OBJECT_UNLOCK (filter);
          if (request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT)) {
            GST_OBJECT_LOCK (filter);
            goto unprotect;
          } else {
            GST_WARNING_OBJECT (filter, "Hard limit reached, no new key, "
                "dropping");
          }
        } else {
          GST_WARNING_OBJECT (filter, "Could not find matching stream, "
              "dropping");
        }
        break;
      case err_status_auth_fail:
        GST_WARNING_OBJECT (filter, "Error authentication packet, dropping");
        break;
      case err_status_cipher_fail:
        GST_WARNING_OBJECT (filter, "Error while decrypting packet, dropping");
        break;
      default:
        GST_WARNING_OBJECT (filter, "Other error, dropping");
        break;
    }

    goto drop_buffer;
  }

  gst_buffer_set_size (buf, size);

  /* If all is well, we may have reached soft limit */
  if (gst_srtp_get_soft_limit_reached ())
    request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);

push_out:
  /* Push buffer to source pad */
  if (is_rtcp) {
    otherpad = filter->rtcp_srcpad;
    if (!filter->rtcp_has_segment)
      gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
          filter->rtp_srcpad, TRUE);
  } else {
    otherpad = filter->rtp_srcpad;
    if (!filter->rtp_has_segment)
      gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
          filter->rtcp_srcpad, FALSE);
  }
  ret = gst_pad_push (otherpad, buf);

  return ret;

drop_buffer:
  /* Drop buffer, except if gst_pad_push returned OK or an error */

  gst_buffer_unref (buf);

  return ret;
}