Index: gst-libs/gst/rtp/gstbasertpaudiopayload.h
===================================================================
--- gst-libs/gst/rtp/gstbasertpaudiopayload.h	(revision 0)
+++ gst-libs/gst/rtp/gstbasertpaudiopayload.h	(revision 0)
@@ -0,0 +1,88 @@
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <burger@speedy.org> 
+ *
+ * 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;
+
+#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
+  (gst_basertpaudiopayload_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,GstBaseRTPAudioPayload))
+#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(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
+
+typedef enum {
+  AUDIO_CODEC_TYPE_NONE,
+  AUDIO_CODEC_TYPE_FRAME_BASED,
+  AUDIO_CODEC_TYPE_SAMPLE_BASED
+} AudioCodecType;
+
+struct _GstBaseRTPAudioPayload
+{
+  GstBaseRTPPayload payload;
+
+  GstClockTime adapter_base_ts;
+  GstAdapter *adapter;
+  gint frame_size;
+  gint frame_duration;
+
+  gint sample_size;
+
+  AudioCodecType type;
+};
+
+struct _GstBaseRTPAudioPayloadClass
+{
+  GstBaseRTPPayloadClass parent_class;
+};
+
+gboolean gst_basertpaudiopayload_plugin_init (GstPlugin * plugin);
+
+GType gst_basertpaudiopayload_get_type (void);
+
+void
+gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
+
+void
+gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
+
+void
+gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint frame_duration, gint frame_size);
+
+void
+gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint sample_size);
+
+G_END_DECLS
+
+#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
Index: gst-libs/gst/rtp/gstrtpbuffer.h
===================================================================
--- gst-libs/gst/rtp/gstrtpbuffer.h	(revision 1633)
+++ gst-libs/gst/rtp/gstrtpbuffer.h	(working copy)
@@ -71,6 +71,8 @@
 #define GST_RTP_PAYLOAD_MPV_STRING "32"
 #define GST_RTP_PAYLOAD_H263_STRING "34"
 
+#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]"
+
 /* creating buffers */
 GstBuffer*      gst_rtp_buffer_new              (void);
 void            gst_rtp_buffer_allocate_data    (GstBuffer *buffer, guint payload_len, 
Index: gst-libs/gst/rtp/Makefile.am
===================================================================
--- gst-libs/gst/rtp/Makefile.am	(revision 1633)
+++ gst-libs/gst/rtp/Makefile.am	(working copy)
@@ -2,12 +2,14 @@
 
 libgstrtpinclude_HEADERS = gstrtpbuffer.h \
 			gstbasertppayload.h \
+			gstbasertpaudiopayload.h \
 			gstbasertpdepayload.h
 
 lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
 
 libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
   				gstbasertppayload.c \
+  				gstbasertpaudiopayload.c \
   				gstbasertpdepayload.c
 
 libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS)
Index: gst-libs/gst/rtp/gstbasertpaudiopayload.c
===================================================================
--- gst-libs/gst/rtp/gstbasertpaudiopayload.c	(revision 0)
+++ gst-libs/gst/rtp/gstbasertpaudiopayload.c	(revision 0)
@@ -0,0 +1,433 @@
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <burger@speedy.org> 
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <math.h>
+
+#include "gstbasertpaudiopayload.h"
+
+GST_DEBUG_CATEGORY (basertpaudiopayload_debug);
+#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
+
+/* let us define a minimum of 10 ms for sample based codecs */
+#define GST_RTP_MIN_PTIME_MS 10
+
+static void
+gst_basertpaudiopayload_finalize (GObject * object);
+
+static GstFlowReturn
+gst_basertpaudiopayload_push (GstBaseRTPPayload *basepayload, guint8 *data, guint
+    payload_len, GstClockTime timestamp);
+
+static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * payload,
+    GstBuffer * buffer);
+
+static GstFlowReturn
+gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer);
+
+static GstFlowReturn
+gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer);
+
+GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload, GstBaseRTPPayload,
+    GST_TYPE_BASE_RTP_PAYLOAD);
+
+static void
+gst_basertpaudiopayload_base_init (gpointer klass)
+{
+}
+
+static void
+gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+  gobject_class->finalize = gst_basertpaudiopayload_finalize;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+  gstbasertppayload_class->handle_buffer = gst_basertpaudiopayload_handle_buffer;
+
+  GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
+      "base audio RTP payloader");
+}
+
+static void
+gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload, GstBaseRTPAudioPayloadClass * klass)
+{
+  basertpaudiopayload->adapter = gst_adapter_new();
+  basertpaudiopayload->adapter_base_ts = 0;
+
+  basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE;
+
+  /* these need to be set by child object if frame based */
+  basertpaudiopayload->frame_size = 0;
+  basertpaudiopayload->frame_duration = 0;
+
+  /* these need to be set by child object if sample based */
+  basertpaudiopayload->sample_size = 0;
+}
+
+static void
+gst_basertpaudiopayload_finalize (GObject * object)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+  g_object_unref (basertpaudiopayload->adapter);
+  basertpaudiopayload->adapter = NULL;
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
+}
+
+void
+gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE)
+  {
+    GST_ERROR_OBJECT (basertpaudiopayload, "Codec type already set! You should only set this once!");
+  }
+  basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED;
+}
+
+void
+gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE)
+  {
+    GST_ERROR_OBJECT (basertpaudiopayload, "Codec type already set! You should only set this once!");
+  }
+  basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
+}
+
+/* These are options that need to be set for frame based audio codecs */
+void
+gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint frame_duration, gint frame_size)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  basertpaudiopayload->frame_size = frame_size;
+  basertpaudiopayload->frame_duration = frame_duration;
+}
+
+void
+gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
+    *basertpaudiopayload, gint sample_size)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  basertpaudiopayload->sample_size = sample_size;
+}
+
+static GstFlowReturn
+gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload *basepayload, GstBuffer
+    *buffer)
+{
+  GstFlowReturn ret;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD(basepayload);
+
+  ret = GST_FLOW_ERROR;
+
+  if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED)
+  {
+    ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload,
+        buffer);
+  }
+  else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED)
+  {
+    ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload,
+        buffer);
+  }
+  else
+  {
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
+  }
+
+  return ret;
+}
+
+/* this assumes all frames have a constant duration and a constant size */
+static GstFlowReturn
+gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  guint payload_len;
+  guint8 *data;
+  GstFlowReturn ret;
+  guint available;
+  gint frame_size, frame_duration;
+
+  guint maxptime_octets = G_MAXUINT;
+
+  ret = GST_FLOW_ERROR;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD(basepayload);
+
+  if (basertpaudiopayload->frame_size == 0 ||
+      basertpaudiopayload->frame_duration == 0)
+  {
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
+    gst_buffer_unref (buffer);
+    return GST_FLOW_ERROR;
+  }
+  frame_size = basertpaudiopayload->frame_size;
+  frame_duration = basertpaudiopayload->frame_duration;
+
+  /* If buffer fits on an RTP packet, let's just push it through without using
+   * the adapter */
+  /* this will check again max_ptime and max_mtu */
+  if (!gst_basertppayload_is_filled (basepayload,
+        gst_rtp_buffer_calc_packet_len(GST_BUFFER_SIZE(buffer), 0, 0),
+          GST_BUFFER_DURATION(buffer)))
+  {
+    ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA(buffer),
+        GST_BUFFER_SIZE(buffer), GST_BUFFER_TIMESTAMP(buffer));
+    gst_buffer_unref (buffer);
+
+    return ret;
+  }
+
+  /* TODO : would be nice if we had some property that told the payloader to put
+   * just 1 frame per RTP packet, for the moment we can set the ptime to 0 or
+   * something smaller or equal to a frame duration */
+
+  /* max number of bytes based on given ptime, has to be multiple of
+   * frame_duration */
+  if (basepayload->max_ptime != -1)
+  {
+    guint ptime_ms = basepayload->max_ptime / 1000000;
+    maxptime_octets = frame_size * (int)(
+        ptime_ms / frame_duration);
+    if (maxptime_octets == 0)
+    {
+      GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than minimum %d ms, overwriting to minimum", 
+          ptime_ms, frame_duration);
+      maxptime_octets = frame_size;
+    }
+  }
+
+  /* if the adapter is empty (should be), let's set the base timestamp */
+  if (gst_adapter_available (basertpaudiopayload->adapter) == 0)
+  {
+    basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP(buffer);
+  }
+  else
+  {
+    GST_ERROR_OBJECT (basertpaudiopayload, "Adapter should be empty but is not!");
+    return GST_FLOW_ERROR;
+  }
+
+  gst_adapter_push (basertpaudiopayload->adapter, buffer);
+
+  available = gst_adapter_available (basertpaudiopayload->adapter);
+
+  /* as long as we have full frames */
+  /* this loop will always empty the adapter till the last frame */
+  /* TODO Make it possible to set a minimum size per packet, this way the
+   * algorithm doesn't empty the adapter if there is too little data left and
+   * will wait until the next buffers to arrive */
+  while (available >= frame_size)
+  {
+    /* we need to see how many frames we can get based on maximum MTU, maximum
+     * ptime and the number of bytes available in the adapter */
+    payload_len = MIN(MIN(
+        /* MTU max */
+        (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+            (basertpaudiopayload), 0, 0) / frame_size) * frame_size,
+        /* ptime max */
+        maxptime_octets),
+        /* currently available */
+        floor (available / frame_size) * frame_size
+        );
+
+    data = (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
+    ret = gst_basertpaudiopayload_push (basepayload, data, payload_len,
+        basertpaudiopayload->adapter_base_ts); 
+
+    gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
+    gfloat ts_inc = (payload_len * frame_duration) / frame_size;
+    ts_inc = ts_inc * GST_MSECOND;
+    basertpaudiopayload->adapter_base_ts += ts_inc;
+    GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc, ts_inc*GST_MSECOND, (payload_len * frame_duration) / frame_size);
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
+        GST_TIME_ARGS(basertpaudiopayload->adapter_base_ts));
+
+    available = gst_adapter_available (basertpaudiopayload->adapter);
+  }
+
+  /* adapter should be freed by now */
+  if (available != 0)
+  {
+    GST_ERROR_OBJECT (basertpaudiopayload, "Adapter should be empty but is not!");
+    return GST_FLOW_ERROR;
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  guint payload_len;
+  guint8 *data;
+  GstFlowReturn ret;
+  guint available;
+
+  guint maxptime_octets = G_MAXUINT;
+
+  guint minptime_octets = 0;
+  guint sample_size;
+
+  ret = GST_FLOW_ERROR;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD(basepayload);
+
+  if (basertpaudiopayload->sample_size == 0)
+  {
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
+    gst_buffer_unref (buffer);
+    return GST_FLOW_ERROR;
+  }
+  sample_size = basertpaudiopayload->sample_size;
+
+  /* If buffer fits on an RTP packet, let's just push it through without using
+   * the adapter */
+  /* this will check again max_ptime and max_mtu */
+  if (!gst_basertppayload_is_filled (basepayload,
+        gst_rtp_buffer_calc_packet_len(GST_BUFFER_SIZE(buffer), 0, 0),
+          GST_BUFFER_DURATION(buffer)))
+  {
+    ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA(buffer),
+        GST_BUFFER_SIZE(buffer), GST_BUFFER_TIMESTAMP(buffer));
+    gst_buffer_unref (buffer);
+
+    return ret;
+  }
+
+  /* max number of bytes based on given ptime */
+  if (basepayload->max_ptime != -1)
+  {
+    maxptime_octets = basepayload->max_ptime * basepayload->clock_rate /
+      (sample_size * GST_SECOND);
+    minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate /
+      (sample_size * 1000);
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Calculated max_octects %u and min_octets %u", maxptime_octets, minptime_octets);
+    if (maxptime_octets < minptime_octets)
+    {
+      GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than minimum %d, replacing by %d",
+          maxptime_octets, minptime_octets, minptime_octets);
+      maxptime_octets = minptime_octets;
+    }
+  }
+
+  /* if the adapter is empty (should be), let's set the base timestamp */
+  if (gst_adapter_available (basertpaudiopayload->adapter) == 0)
+  {
+    basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP(buffer);
+    GST_DEBUG_OBJECT(basertpaudiopayload, "Setting to %" GST_TIME_FORMAT,
+        GST_TIME_ARGS(GST_BUFFER_TIMESTAMP(buffer)));
+  }
+
+  gst_adapter_push (basertpaudiopayload->adapter, buffer);
+
+  available = gst_adapter_available (basertpaudiopayload->adapter);
+
+  /* as long as we have full frames */
+  /* this loop will always empty the adapter till the last frame */
+  /* TODO Make it possible to set a minimum size per packet, this way the
+   * algorithm doesn't empty the adapter if there is too little data left and
+   * will wait until the next buffers to arrive */
+  while (available >= minptime_octets)
+  {
+    /* we need to see how many frames we can get based on maximum MTU, maximum
+     * ptime and the number of bytes available in the adapter */
+    payload_len = MIN(MIN(
+        /* MTU max */
+        gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+            (basertpaudiopayload), 0, 0),
+        /* ptime max */
+        maxptime_octets),
+        /* currently available */
+        available
+        );
+
+    data = (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
+    GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
+        GST_TIME_ARGS(basertpaudiopayload->adapter_base_ts));
+    ret = gst_basertpaudiopayload_push (basepayload, data, payload_len,
+        basertpaudiopayload->adapter_base_ts); 
+
+    gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
+    gfloat num = payload_len;
+    gfloat datarate = (sample_size * basepayload->clock_rate);
+    basertpaudiopayload->adapter_base_ts += 
+      /* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */
+      num / datarate * GST_SECOND;
+    GST_DEBUG_OBJECT(basertpaudiopayload, "Calculating ts inc %f %f %f", num, datarate, num/datarate * GST_SECOND);
+    GST_DEBUG_OBJECT(basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
+        GST_TIME_ARGS(basertpaudiopayload->adapter_base_ts));
+
+    available = gst_adapter_available (basertpaudiopayload->adapter);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_basertpaudiopayload_push (GstBaseRTPPayload *basepayload, guint8 *data, guint
+    payload_len, GstClockTime timestamp)
+{
+  GstBuffer *outbuf;
+  guint8 *payload;
+  GstFlowReturn ret;
+
+  /* create buffer to hold the payload */
+  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+  /* copy payload */
+  gst_rtp_buffer_set_payload_type (outbuf,
+      basepayload->pt);
+  payload = gst_rtp_buffer_get_payload (outbuf);
+  memcpy (payload, data, payload_len);
+
+  GST_BUFFER_TIMESTAMP (outbuf) = timestamp; 
+  ret = gst_basertppayload_push (basepayload, outbuf);
+
+  return ret;
+}
