gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode...

Original commit message from CVS:
Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
(gst_wavparse_stream_data):
Fix massive memory leak when operating in streaming mode due to
GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
Fixes #407057.

diff --git a/ChangeLog b/ChangeLog
index 0bc4b09..70016c9 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,13 @@
+2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>
+
+       Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>
+
+       * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
+       (gst_wavparse_stream_data):
+         Fix massive memory leak when operating in streaming mode due to
+         GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
+         Fixes #407057.
+
 2007-02-12  Stefan Kost  <ensonic@users.sf.net>
 
        * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),

diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index e6a1448..72f0e94 100644 (file)
--- a/gst/wavparse/gstwavparse.c
+++ b/gst/wavparse/gstwavparse.c
@@ -1319,12 +1319,10 @@ static GstFlowReturn
 gst_wavparse_parse_stream_init (GstWavParse * wav)
 {
   if (gst_adapter_available (wav->adapter) >= 12) {
-    GstBuffer *tmp = gst_buffer_new ();
+    GstBuffer *tmp;
 
     /* _take flushes the data */
-    GST_BUFFER_DATA (tmp) = gst_adapter_take (wav->adapter, 12);
-    GST_BUFFER_SIZE (tmp) = 12;
-
+    tmp = gst_adapter_take_buffer (wav->adapter, 12);
     GST_DEBUG ("Parsing wav header");
     if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) {
       return GST_FLOW_ERROR;
@@ -1458,9 +1456,7 @@ iterate_adapter:
       return GST_FLOW_OK;
     }
 
-    buf = gst_buffer_new ();
-    GST_BUFFER_DATA (buf) = gst_adapter_take (wav->adapter, desired);
-    GST_BUFFER_SIZE (buf) = desired;
+    buf = gst_adapter_take_buffer (wav->adapter, desired);
   } else {
     if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
                 desired, &buf)) != GST_FLOW_OK)