--- /dev/null
+/* 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__ */