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, 35 insertions, 2 deletions

diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c
index 2b8b8652..92d72240 100644
--- a/gst/rtp/gstrtpmp4gpay.c
+++ b/gst/rtp/gstrtpmp4gpay.c
@@ -85,6 +85,8 @@ static void gst_rtp_mp4g_pay_finalize (GObject * object);
 
 static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
     GstCaps * caps);
+static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
+    GstStateChange transition);
 static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
     payload, GstBuffer * buffer);
 
@@ -143,6 +145,8 @@ gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass)
 
   gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
 
+  gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
+
   gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
 
@@ -171,6 +175,10 @@ gst_rtp_mp4g_pay_finalize (GObject * object)
   g_free (rtpmp4gpay->params);
   rtpmp4gpay->params = NULL;
 
+  if (rtpmp4gpay->config)
+    gst_buffer_unref (rtpmp4gpay->config);
+  rtpmp4gpay->config = NULL;
+
   g_free (rtpmp4gpay->profile);
   rtpmp4gpay->profile = NULL;
 
@@ -504,7 +512,8 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
     /* marker only if the packet is complete */
     gst_rtp_buffer_set_marker (outbuf, avail <= payload_len);
 
-    GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;
+    GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
+    GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
 
     ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
 
@@ -528,7 +537,8 @@ gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
   rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
 
-  rtpmp4gpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+  rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer);
 
   /* we always encode and flush a full AU */
   gst_adapter_push (rtpmp4gpay->adapter, buffer);
@@ -537,6 +547,29 @@ gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
   return ret;
 }
 
+static GstStateChangeReturn
+gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRtpMP4GPay *rtpmp4gpay;
+  GstStateChangeReturn ret;
+
+  rtpmp4gpay = GST_RTP_MP4G_PAY (element);
+
+  switch (transition) {
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    default:
+      break;
+  }
+  return ret;
+}
+
+
 gboolean
 gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
 {