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Code Issues 1 Projects Releases Wiki Activity

srtpdec: Separate buffer encoding functionality into a different function

Browse source 
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
1 changed files with 52 additions and 30 deletions
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82
ext/srtp/gstsrtpdec.c
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,
gboolean is_rtcp)
/*
* 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)
{
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;
}
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),
@ -1038,11 +1021,11 @@ gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
/* Change buffer to remove protection */
buf = gst_buffer_make_writable (buf);
unprotect:
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
size = map.size;
unprotect:
gst_srtp_init_event_reporter ();
if (is_rtcp)
@ -1074,8 +1057,6 @@ unprotect:
err = srtp_unprotect (filter->session, map.data, &size);
}
gst_buffer_unmap (buf, &map);
GST_OBJECT_UNLOCK (filter);
if (err != err_status_ok) {
@ -1113,10 +1094,51 @@ unprotect:
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;
}
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 (gst_srtp_get_soft_limit_reached ())