srtpdec: Separate buffer encoding functionality into a different function

https://bugzilla.gnome.org/show_bug.cgi?id=746387
This commit is contained in:
Jose Antonio Santos Cadenas 2015-03-18 10:47:15 +01:00 committed by Sebastian Dröge
parent f295beda07
commit 8d2e98bc3f

View File

@ -1003,33 +1003,16 @@ gst_srtp_dec_push_early_events (GstSrtpDec * filter, GstPad * pad,
} }
static GstFlowReturn /*
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf, * This function should be called while holding the filter lock
gboolean is_rtcp) */
static gboolean
gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf,
gboolean is_rtcp, guint32 ssrc)
{ {
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; GstMapInfo map;
err_status_t err;
GST_OBJECT_LOCK (filter); gint size;
/* 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 GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT
" with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf), " with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf),
@ -1038,11 +1021,11 @@ gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
/* Change buffer to remove protection */ /* Change buffer to remove protection */
buf = gst_buffer_make_writable (buf); buf = gst_buffer_make_writable (buf);
unprotect:
gst_buffer_map (buf, &map, GST_MAP_READWRITE); gst_buffer_map (buf, &map, GST_MAP_READWRITE);
size = map.size; size = map.size;
unprotect:
gst_srtp_init_event_reporter (); gst_srtp_init_event_reporter ();
if (is_rtcp) if (is_rtcp)
@ -1074,8 +1057,6 @@ unprotect:
err = srtp_unprotect (filter->session, map.data, &size); err = srtp_unprotect (filter->session, map.data, &size);
} }
gst_buffer_unmap (buf, &map);
GST_OBJECT_UNLOCK (filter); GST_OBJECT_UNLOCK (filter);
if (err != err_status_ok) { if (err != err_status_ok) {
@ -1113,10 +1094,51 @@ unprotect:
break; 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;
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; goto drop_buffer;
} }
gst_buffer_set_size (buf, size); 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 all is well, we may have reached soft limit */
if (gst_srtp_get_soft_limit_reached ()) if (gst_srtp_get_soft_limit_reached ())