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-rw-r--r--ext/audiofile/gstafparse.c10
-rw-r--r--ext/audiofile/gstafsink.c5
-rw-r--r--ext/audiofile/gstafsrc.c8
3 files changed, 12 insertions, 11 deletions
diff --git a/ext/audiofile/gstafparse.c b/ext/audiofile/gstafparse.c
index 183f5147..248389f6 100644
--- a/ext/audiofile/gstafparse.c
+++ b/ext/audiofile/gstafparse.c
@@ -195,7 +195,7 @@ gst_afparse_loop(GstElement *element)
/* just stop if we cannot open the file */
if (!gst_afparse_open_file (afparse)){
gst_bytestream_destroy ((GstByteStream *) afparse->vfile->closure);
- gst_pad_push (afparse->srcpad, GST_BUFFER(gst_event_new (GST_EVENT_EOS)));
+ gst_pad_push (afparse->srcpad, GST_DATA(gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
return;
}
@@ -236,14 +236,14 @@ gst_afparse_loop(GstElement *element)
gst_bytestream_get_status (bs, &waiting, &event);
if (event && GST_EVENT_TYPE(event) == GST_EVENT_EOS) {
gst_pad_push (afparse->srcpad,
- GST_BUFFER (gst_event_new (GST_EVENT_EOS)));
+ GST_DATA (gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
break;
}
}
else {
GST_BUFFER_TIMESTAMP(buf) = afparse->timestamp;
- gst_pad_push (afparse->srcpad, buf);
+ gst_pad_push (afparse->srcpad, GST_DATA (buf));
if (got_bytes != bytes_per_read){
/* this shouldn't happen very often */
/* FIXME calculate the timestamps based on the fewer bytes received */
@@ -269,12 +269,12 @@ gst_afparse_loop(GstElement *element)
if (numframes < 1){
gst_buffer_unref(buf);
- gst_pad_push (afparse->srcpad, GST_BUFFER(gst_event_new (GST_EVENT_EOS)));
+ gst_pad_push (afparse->srcpad, GST_DATA(gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
break;
}
GST_BUFFER_SIZE(buf) = numframes * frames_to_bytes;
- gst_pad_push (afparse->srcpad, buf);
+ gst_pad_push (afparse->srcpad, GST_DATA (buf));
afparse->timestamp += numframes * 1E9 / afparse->rate;
}
while (TRUE);
diff --git a/ext/audiofile/gstafsink.c b/ext/audiofile/gstafsink.c
index 3392717c..fd80975e 100644
--- a/ext/audiofile/gstafsink.c
+++ b/ext/audiofile/gstafsink.c
@@ -104,7 +104,7 @@ static void gst_afsink_init (GstAFSink *afsink);
static gboolean gst_afsink_open_file (GstAFSink *sink);
static void gst_afsink_close_file (GstAFSink *sink);
-static void gst_afsink_chain (GstPad *pad,GstBuffer *buf);
+static void gst_afsink_chain (GstPad *pad,GstData *_data);
static void gst_afsink_set_property (GObject *object, guint prop_id, const GValue *value,
GParamSpec *pspec);
@@ -389,8 +389,9 @@ gst_afsink_close_file (GstAFSink *sink)
* take the buffer from the pad and write to file if it's open
*/
static void
-gst_afsink_chain (GstPad *pad, GstBuffer *buf)
+gst_afsink_chain (GstPad *pad, GstData *_data)
{
+ GstBuffer *buf = GST_BUFFER (_data);
GstAFSink *afsink;
int ret = 0;
diff --git a/ext/audiofile/gstafsrc.c b/ext/audiofile/gstafsrc.c
index 2b817310..8c1a0c69 100644
--- a/ext/audiofile/gstafsrc.c
+++ b/ext/audiofile/gstafsrc.c
@@ -105,7 +105,7 @@ static void gst_afsrc_init (GstAFSrc *afsrc);
static gboolean gst_afsrc_open_file (GstAFSrc *src);
static void gst_afsrc_close_file (GstAFSrc *src);
-static GstBuffer* gst_afsrc_get (GstPad *pad);
+static GstData* gst_afsrc_get (GstPad *pad);
static void gst_afsrc_set_property (GObject *object, guint prop_id,
const GValue *value, GParamSpec *pspec);
@@ -190,7 +190,7 @@ gst_afsrc_init (GstAFSrc *afsrc)
afsrc->framestamp = 0;
}
-static GstBuffer *
+static GstData *
gst_afsrc_get (GstPad *pad)
{
GstAFSrc *src;
@@ -216,7 +216,7 @@ gst_afsrc_get (GstPad *pad)
readbytes = readframes * (src->channels * src->width / 8);
if (readbytes == 0) {
gst_element_set_eos (GST_ELEMENT (src));
- return GST_BUFFER (gst_event_new (GST_EVENT_EOS));
+ return GST_DATA (gst_event_new (GST_EVENT_EOS));
}
GST_BUFFER_SIZE (buf) = readbytes;
@@ -231,7 +231,7 @@ gst_afsrc_get (GstPad *pad)
GST_BUFFER_TIMESTAMP (buf) / 1E9);
/* g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes); */
- return buf;
+ return GST_DATA (buf);
}
static void