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;
}
static gboolean
gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean ret;
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, TRUE);
gst_event_unref (event);
return ret;
case GST_EVENT_SEGMENT:
/* Make sure to send a caps event downstream before the segment event,
* even if upstream didn't */
if (!gst_pad_has_current_caps (filter->rtcp_srcpad)) {
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
gst_pad_set_caps (filter->rtcp_srcpad, caps);
gst_caps_unref (caps);
}
filter->rtcp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtcp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_srtp_dec_sink_setcaps (GstPad * pad, GstObject * parent,
GstCaps * caps, gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstStructure *ps;
gboolean ret = FALSE;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
ps = gst_caps_get_structure (caps, 0);
if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT) &&
gst_structure_has_field_typed (ps, "roc", G_TYPE_UINT) &&
gst_structure_has_field_typed (ps, "srtp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtp-auth", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-auth", G_TYPE_STRING)) {
guint ssrc;
gst_structure_get_uint (ps, "ssrc", &ssrc);
if (!update_session_stream_from_caps (filter, ssrc, caps)) {
GST_WARNING_OBJECT (pad, "Could not create session from pad caps: %"
GST_PTR_FORMAT, caps);
return FALSE;
}
}
caps = gst_caps_copy (caps);
ps = gst_caps_get_structure (caps, 0);
gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth",
"srtcp-cipher", "srtcp-auth", NULL);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-rtcp");
else
gst_structure_set_name (ps, "application/x-rtp");
otherpad = gst_pad_get_element_private (pad);
ret = gst_pad_set_caps (otherpad, caps);
gst_caps_unref (caps);
return ret;
}
static void
gst_srtp_dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSrtpDec *filter = GST_SRTP_DEC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_REPLAY_WINDOW_SIZE:
g_value_set_uint (value, filter->replay_window_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static GstStateChangeReturn
gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstSrtpDec *filter;
filter = GST_SRTP_DEC (element);
GST_OBJECT_LOCK (filter);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->streams = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) free_stream);
filter->rtp_has_segment = FALSE;
filter->rtcp_has_segment = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
GST_OBJECT_UNLOCK (filter);
res = GST_ELEMENT_CLASS (gst_srtp_dec_parent_class)->change_state (element,
transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_srtp_dec_clear_streams (filter);
g_hash_table_unref (filter->streams);
filter->streams = NULL;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
static gboolean
gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
return gst_srtp_dec_sink_setcaps (pad, parent, caps, FALSE);
case GST_EVENT_SEGMENT:
filter->rtp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static GstIterator *
gst_srtp_dec_iterate_internal_links (GstPad * pad, GstObject * parent,
gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad = NULL;
GstIterator *it = NULL;
otherpad = (GstPad *) gst_pad_get_element_private (pad);
if (otherpad) {
GValue val = { 0 };
g_value_init (&val, GST_TYPE_PAD);
g_value_set_object (&val, otherpad);
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
} else {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), CORE, PAD, (NULL),
("Unable to get linked pad"));
}
return it;
}
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;
}
