rtpL16depay: don't crash if data is not modulo channels*width

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpL16depay.c

diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c
index a842695..3695da9 100644 (file)
--- a/gst/rtp/gstrtpL16depay.c
+++ b/gst/rtp/gstrtpL16depay.c
@@ -13,8 +13,24 @@
  *
  * You should have received a copy of the GNU Library General Public
  * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpL16depay
+ * @see_also: rtpL16pay
+ *
+ * Extract raw audio from RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * <refsect2>
+ * <title>Example pipeline</title>
+ * |[
+ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
+ * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
+ * the rtpL16pay example to create the RTP stream.
+ * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
@@ -28,6 +44,7 @@
 
 #include "gstrtpL16depay.h"
 #include "gstrtpchannels.h"
+#include "gstrtputils.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
 #define GST_CAT_DEFAULT (rtpL16depay_debug)
@@ -47,9 +64,7 @@ static GstStaticPadTemplate gst_rtp_L16_depay_sink_template =
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
-        "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) [ 1, MAX ], "
+        "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
         /* "channels = (int) [1, MAX]"  */
         /* "emphasis = (string) ANY" */
         /* "channel-order = (string) ANY" */
@@ -70,7 +85,7 @@ G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
 static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
     GstCaps * caps);
 static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
-    GstBuffer * buf);
+    GstRTPBuffer * rtp);
 
 static void
 gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
@@ -82,12 +97,12 @@ gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
   gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
 
   gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
-  gstrtpbasedepayload_class->process = gst_rtp_L16_depay_process;
+  gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
 
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_L16_depay_src_template));
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template));
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_L16_depay_src_template);
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_L16_depay_sink_template);
 
   gst_element_class_set_static_metadata (gstelement_class,
       "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
@@ -101,7 +116,6 @@ gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
 static void
 gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
 {
-  /* needed because of GST_BOILERPLATE */
 }
 
 static gint
@@ -212,41 +226,45 @@ no_clockrate:
 }
 
 static GstBuffer *
-gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
+gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
 {
   GstRtpL16Depay *rtpL16depay;
   GstBuffer *outbuf;
   gint payload_len;
   gboolean marker;
-  GstRTPBuffer rtp = { NULL };
+  GstAudioInfo *info;
 
   rtpL16depay = GST_RTP_L16_DEPAY (depayload);
 
-  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
-  payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+  payload_len = gst_rtp_buffer_get_payload_len (rtp);
 
   if (payload_len <= 0)
     goto empty_packet;
 
   GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
 
-  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
-  marker = gst_rtp_buffer_get_marker (&rtp);
+  outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
+  marker = gst_rtp_buffer_get_marker (rtp);
 
   if (marker) {
-    /* mark talk spurt with DISCONT */
-    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+    /* mark talk spurt with RESYNC */
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
   }
 
   outbuf = gst_buffer_make_writable (outbuf);
+  info = &rtpL16depay->info;
+
+  if (payload_len % info->bpf != 0)
+    goto wrong_payload_size;
+
   if (rtpL16depay->order &&
       !gst_audio_buffer_reorder_channels (outbuf,
-          rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
-          rtpL16depay->info.position, rtpL16depay->order->pos)) {
+          info->finfo->format, info->channels,
+          info->position, rtpL16depay->order->pos)) {
     goto reorder_failed;
   }
 
-  gst_rtp_buffer_unmap (&rtp);
+  gst_rtp_drop_non_audio_meta (rtpL16depay, outbuf);
 
   return outbuf;
 
@@ -255,14 +273,18 @@ empty_packet:
   {
     GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
         ("Empty Payload."), (NULL));
-    gst_rtp_buffer_unmap (&rtp);
+    return NULL;
+  }
+wrong_payload_size:
+  {
+    GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
+        ("Wrong Payload Size."), (NULL));
     return NULL;
   }
 reorder_failed:
   {
     GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
         ("Channel reordering failed."), (NULL));
-    gst_rtp_buffer_unmap (&rtp);
     return NULL;
   }
 }