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; }
/* Return a stream structure for a given buffer */ static GstSrtpDecSsrcStream * validate_buffer (GstSrtpDec * filter, GstBuffer * buf, guint32 * ssrc, gboolean * is_rtcp) { GstSrtpDecSsrcStream *stream = NULL; GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT; if (gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf)) { if (gst_rtp_buffer_get_payload_type (&rtpbuf) < 64 || gst_rtp_buffer_get_payload_type (&rtpbuf) > 80) { *ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf); gst_rtp_buffer_unmap (&rtpbuf); *is_rtcp = FALSE; goto have_ssrc; } gst_rtp_buffer_unmap (&rtpbuf); } if (rtcp_buffer_get_ssrc (buf, ssrc)) { *is_rtcp = TRUE; } else { GST_WARNING_OBJECT (filter, "No SSRC found in buffer"); return NULL; } have_ssrc: stream = find_stream_by_ssrc (filter, *ssrc); if (stream) return stream; return request_key_with_signal (filter, *ssrc, SIGNAL_REQUEST_KEY); }
/* * This function should be called while holding the filter lock */ static gboolean gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf, gboolean is_rtcp, guint32 ssrc) { GstMapInfo map; err_status_t err; gint size; 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); gst_buffer_map (buf, &map, GST_MAP_READWRITE); size = map.size; unprotect: 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 | GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING, &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_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; } gst_buffer_unmap (buf, &map); GST_OBJECT_LOCK (filter); return FALSE; } gst_buffer_unmap (buf, &map); gst_buffer_set_size (buf, size); GST_OBJECT_LOCK (filter); return TRUE; }
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; }