fixes: #514889

Original commit message from CVS:
patch by:  Wim Taymans  <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps

1 files changed, 48 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c
index 30f484b2..f9fa15b2 100644
--- a/gst/rtp/gstrtpvorbispay.c
+++ b/gst/rtp/gstrtpvorbispay.c
@@ -75,6 +75,8 @@ GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload,
 
 static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload,
     GstCaps * caps);
+static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement *
+    element, GstStateChange transition);
 static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad,
     GstBuffer * buffer);
 
@@ -104,6 +106,8 @@ gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass)
 
   parent_class = g_type_class_peek_parent (klass);
 
+  gstelement_class->change_state = gst_rtp_vorbis_pay_change_state;
+
   gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
 
@@ -118,6 +122,18 @@ gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay,
   /* needed because of GST_BOILERPLATE */
 }
 
+static void
+gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay)
+{
+  g_list_foreach (rtpvorbispay->headers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (rtpvorbispay->headers);
+  rtpvorbispay->headers = NULL;
+
+  if (rtpvorbispay->packet)
+    gst_buffer_unref (rtpvorbispay->packet);
+  rtpvorbispay->packet = NULL;
+}
+
 static gboolean
 gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
 {
@@ -623,6 +639,8 @@ gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
         rtpvorbispay->payload_duration += duration;
     }
   }
+  gst_buffer_unref (buffer);
+
 done:
   return ret;
 
@@ -631,26 +649,56 @@ wrong_size:
   {
     GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
         ("Invalid packet size (1 < %d <= 0xffff)", size), (NULL));
+    gst_buffer_unref (buffer);
     return GST_FLOW_OK;
   }
 parse_id_failed:
   {
+    gst_buffer_unref (buffer);
     return GST_FLOW_ERROR;
   }
 unknown_header:
   {
     GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
         (NULL), ("Ignoring unknown header received"));
+    gst_buffer_unref (buffer);
     return GST_FLOW_OK;
   }
 header_error:
   {
     GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
         (NULL), ("Error initializing header config"));
+    gst_buffer_unref (buffer);
     return GST_FLOW_OK;
   }
 }
 
+static GstStateChangeReturn
+gst_rtp_vorbis_pay_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstRtpVorbisPay *rtpvorbispay;
+  GstStateChangeReturn ret;
+
+  rtpvorbispay = GST_RTP_VORBIS_PAY (element);
+
+  switch (transition) {
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_rtp_vorbis_pay_cleanup (rtpvorbispay);
+      break;
+    default:
+      break;
+  }
+  return ret;
+}
+
 gboolean
 gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
 {