--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/rtp/gstbasertpaudiopayload.h	Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,133 @@
+/* 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))
+
+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];
+};
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GType gst_base_rtp_audio_payload_get_type (void);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void
+gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload
+    *basertpaudiopayload);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void
+gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload
+    *basertpaudiopayload);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void
+gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint frame_duration, gint frame_size);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void
+gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint sample_size);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void
+gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint sample_size);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GstFlowReturn
+gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, 
+    const guint8 * data, guint payload_len, GstClockTime timestamp);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GstAdapter*
+gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload 
+    *basertpaudiopayload);
+
+G_END_DECLS
+
+#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */