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diff --git a/gst-plugins-base-subtitles0.10/gst-libs/gst/rtp/gstbasertpaudiopayload.h b/gst-plugins-base-subtitles0.10/gst-libs/gst/rtp/gstbasertpaudiopayload.h
new file mode 100644
index 0000000..3fdb488
--- /dev/null
+++ b/gst-plugins-base-subtitles0.10/gst-libs/gst/rtp/gstbasertpaudiopayload.h
@@ -0,0 +1,98 @@
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * 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.
+ */
+
+#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__
+#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertppayload.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
+typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
+
+typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate;
+
+#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
+  (gst_base_rtp_audio_payload_get_type())
+#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj), \
+  GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
+#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass), \
+  GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayloadClass))
+#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
+#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
+#define GST_BASE_RTP_AUDIO_PAYLOAD_CAST(obj) \
+  ((GstBaseRTPAudioPayload *) (obj))
+
+struct _GstBaseRTPAudioPayload
+{
+  GstBaseRTPPayload payload;
+
+  GstBaseRTPAudioPayloadPrivate *priv;
+
+  GstClockTime base_ts;
+  gint frame_size;
+  gint frame_duration;
+
+  gint sample_size;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstBaseRTPAudioPayloadClass
+{
+  GstBaseRTPPayloadClass parent_class;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_base_rtp_audio_payload_get_type (void);
+
+/* configure frame based */
+void            gst_base_rtp_audio_payload_set_frame_based        (GstBaseRTPAudioPayload *basertpaudiopayload);
+
+void            gst_base_rtp_audio_payload_set_frame_options      (GstBaseRTPAudioPayload *basertpaudiopayload,
+                                                                   gint frame_duration, gint frame_size);
+
+/* configure sample based */
+void            gst_base_rtp_audio_payload_set_sample_based       (GstBaseRTPAudioPayload *basertpaudiopayload);
+void            gst_base_rtp_audio_payload_set_sample_options     (GstBaseRTPAudioPayload *basertpaudiopayload,
+                                                                   gint sample_size);
+void            gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload,
+                                                                   gint sample_size);
+
+/* get the internal adapter */
+GstAdapter*     gst_base_rtp_audio_payload_get_adapter            (GstBaseRTPAudioPayload *basertpaudiopayload);
+
+/* push and flushing data */
+GstFlowReturn   gst_base_rtp_audio_payload_push                   (GstBaseRTPAudioPayload * baseaudiopayload,
+                                                                   const guint8 * data, guint payload_len,
+                                                                   GstClockTime timestamp);
+GstFlowReturn   gst_base_rtp_audio_payload_flush                  (GstBaseRTPAudioPayload * baseaudiopayload,
+                                                                   guint payload_len, GstClockTime timestamp);
+
+G_END_DECLS
+
+#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */