--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/rtp/gstbasertpaudiopayload.c	Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,728 @@
+/* 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.
+ */
+
+/**
+ * SECTION:gstbasertpaudiopayload
+ * @short_description: Base class for audio RTP payloader
+ *
+ * <refsect2>
+ * <para>
+ * Provides a base class for audio RTP payloaders for frame or sample based
+ * audio codecs (constant bitrate)
+ * </para>
+ * <para>
+ * This class derives from GstBaseRTPPayload. It can be used for payloading
+ * audio codecs. It will only work with constant bitrate codecs. It supports
+ * both frame based and sample based codecs. It takes care of packing up the
+ * audio data into RTP packets and filling up the headers accordingly. The
+ * payloading is done based on the maximum MTU (mtu) and the maximum time per
+ * packet (max-ptime). The general idea is to divide large data buffers into
+ * smaller RTP packets. The RTP packet size is the minimum of either the MTU,
+ * max-ptime (if set) or available data. The RTP packet size is always larger or
+ * equal to min-ptime (if set). If min-ptime is not set, any residual data is
+ * sent in a last RTP packet. In the case of frame based codecs, the resulting
+ * RTP packets always contain full frames.
+ * </para>
+ * <title>Usage</title>
+ * <para>
+ * To use this base class, your child element needs to call either
+ * gst_base_rtp_audio_payload_set_frame_based() or
+ * gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the
+ * element's _init() function. Then, the child element must call either
+ * gst_base_rtp_audio_payload_set_frame_options(),
+ * gst_base_rtp_audio_payload_set_sample_options() or
+ * gst_base_rtp_audio_payload_set_samplebits_options. Since
+ * GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element
+ * must set any variables or call/override any functions required by that base
+ * class. The child element does not need to override any other functions
+ * specific to GstBaseRTPAudioPayload.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstbasertpaudiopayload.h"
+
+GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
+#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
+
+typedef enum
+{
+  AUDIO_CODEC_TYPE_NONE,
+  AUDIO_CODEC_TYPE_FRAME_BASED,
+  AUDIO_CODEC_TYPE_SAMPLE_BASED
+} AudioCodecType;
+
+struct _GstBaseRTPAudioPayloadPrivate
+{
+  AudioCodecType type;
+  GstAdapter *adapter;
+  guint64 min_ptime;
+};
+
+
+#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \
+  (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \
+                                GstBaseRTPAudioPayloadPrivate))
+
+static void gst_base_rtp_audio_payload_finalize (GObject * object);
+
+static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
+    * payload, GstBuffer * buffer);
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
+    basepayload, GstBuffer * buffer);
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
+    basepayload, GstBuffer * buffer);
+
+static GstStateChangeReturn
+gst_base_rtp_payload_audio_change_state (GstElement * element,
+    GstStateChange transition);
+static gboolean
+gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event);
+
+GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
+    GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+
+static void
+gst_base_rtp_audio_payload_base_init (gpointer klass)
+{
+}
+
+static void
+gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate));
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  gobject_class->finalize =
+      GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_finalize);
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state);
+
+  gstbasertppayload_class->handle_buffer =
+      GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer);
+  gstbasertppayload_class->handle_event =
+      GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event);
+
+  GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
+      "base audio RTP payloader");
+}
+
+static void
+gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
+    GstBaseRTPAudioPayloadClass * klass)
+{
+  basertpaudiopayload->priv =
+      GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);
+
+  basertpaudiopayload->base_ts = 0;
+
+  basertpaudiopayload->priv->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;
+
+  basertpaudiopayload->priv->adapter = gst_adapter_new ();
+}
+
+static void
+gst_base_rtp_audio_payload_finalize (GObject * object)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+
+  g_object_unref (basertpaudiopayload->priv->adapter);
+
+  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
+}
+
+/**
+ * gst_base_rtp_audio_payload_set_frame_based:
+ * @basertpaudiopayload: a pointer to the element.
+ *
+ * Tells #GstBaseRTPAudioPayload that the child element is for a frame based
+ * audio codec
+ *
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
+    basertpaudiopayload)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
+
+  basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED;
+}
+
+/**
+ * gst_base_rtp_audio_payload_set_sample_based:
+ * @basertpaudiopayload: a pointer to the element.
+ *
+ * Tells #GstBaseRTPAudioPayload that the child element is for a sample based
+ * audio codec
+ *
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
+    basertpaudiopayload)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
+
+  basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
+}
+
+/**
+ * gst_base_rtp_audio_payload_set_frame_options:
+ * @basertpaudiopayload: a pointer to the element.
+ * @frame_duration: The duraction of an audio frame in milliseconds.
+ * @frame_size: The size of an audio frame in bytes.
+ *
+ * Sets the options for frame based audio codecs.
+ *
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_rtp_audio_payload_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;
+
+  if (basertpaudiopayload->priv->adapter) {
+    gst_adapter_clear (basertpaudiopayload->priv->adapter);
+  }
+}
+
+/**
+ * gst_base_rtp_audio_payload_set_sample_options:
+ * @basertpaudiopayload: a pointer to the element.
+ * @sample_size: Size per sample in bytes.
+ *
+ * Sets the options for sample based audio codecs.
+ *
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
+    * basertpaudiopayload, gint sample_size)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  /* sample_size is in bits internally */
+  basertpaudiopayload->sample_size = sample_size * 8;
+
+  if (basertpaudiopayload->priv->adapter) {
+    gst_adapter_clear (basertpaudiopayload->priv->adapter);
+  }
+}
+
+/**
+ * gst_base_rtp_audio_payload_set_samplebits_options:
+ * @basertpaudiopayload: a pointer to the element.
+ * @sample_size: Size per sample in bits.
+ *
+ * Sets the options for sample based audio codecs.
+ *
+ * Since: 0.10.18
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
+    * basertpaudiopayload, gint sample_size)
+{
+  g_return_if_fail (basertpaudiopayload != NULL);
+
+  basertpaudiopayload->sample_size = sample_size;
+
+  if (basertpaudiopayload->priv->adapter) {
+    gst_adapter_clear (basertpaudiopayload->priv->adapter);
+  }
+}
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstFlowReturn ret;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
+
+  ret = GST_FLOW_ERROR;
+
+  if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
+    ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,
+        buffer);
+  } else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
+    ret = gst_base_rtp_audio_payload_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_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
+    basepayload, GstBuffer * buffer)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  guint payload_len;
+  const guint8 *data = NULL;
+  GstFlowReturn ret;
+  guint available;
+  gint frame_size, frame_duration;
+
+  guint maxptime_octets = G_MAXUINT;
+  guint minptime_octets = 0;
+  guint min_payload_len;
+  guint max_payload_len;
+  gboolean use_adapter = FALSE;
+  guint minptime_ms;
+
+  ret = GST_FLOW_OK;
+
+  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;
+
+  /* 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;
+    }
+  }
+
+  max_payload_len = 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);
+
+  /* min number of bytes based on a given ptime, has to be a multiple
+     of frame duration */
+  minptime_ms = basepayload->min_ptime / 1000000;
+
+  minptime_octets = frame_size * (int) (minptime_ms / frame_duration);
+
+  min_payload_len = MAX (minptime_octets, frame_size);
+
+  if (min_payload_len > max_payload_len) {
+    min_payload_len = max_payload_len;
+  }
+
+  GST_DEBUG_OBJECT (basertpaudiopayload,
+      "Calculated min_payload_len %u and max_payload_len %u",
+      min_payload_len, max_payload_len);
+
+  if (basertpaudiopayload->priv->adapter &&
+      gst_adapter_available (basertpaudiopayload->priv->adapter)) {
+    /* If there is always data in the adapter, we have to use it */
+    gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
+    available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+    use_adapter = TRUE;
+  } else {
+    /* let's set the base timestamp */
+    basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
+
+    /* If buffer fits on an RTP packet, let's just push it through */
+    /* this will check against max_ptime and max_mtu */
+    if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
+        GST_BUFFER_SIZE (buffer) <= max_payload_len) {
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+          GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
+          GST_BUFFER_TIMESTAMP (buffer));
+      gst_buffer_unref (buffer);
+
+      return ret;
+    }
+
+    available = GST_BUFFER_SIZE (buffer);
+    data = (guint8 *) GST_BUFFER_DATA (buffer);
+  }
+
+  /* as long as we have full frames */
+  while (available >= min_payload_len) {
+    gfloat ts_inc;
+
+    /* We send as much as we can */
+    payload_len = MIN (max_payload_len, (available / frame_size) * frame_size);
+
+    if (use_adapter) {
+      data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
+    }
+
+    ret =
+        gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
+        basertpaudiopayload->base_ts);
+
+    ts_inc = (payload_len * frame_duration) / frame_size;
+
+    ts_inc = ts_inc * GST_MSECOND;
+    basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc);
+
+    if (use_adapter) {
+      gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
+      available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+    } else {
+      available -= payload_len;
+      data += payload_len;
+    }
+  }
+
+  if (!use_adapter) {
+    if (available != 0 && basertpaudiopayload->priv->adapter) {
+      GstBuffer *buf;
+
+      buf = gst_buffer_create_sub (buffer,
+          GST_BUFFER_SIZE (buffer) - available, available);
+      gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
+    }
+    gst_buffer_unref (buffer);
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
+    basepayload, GstBuffer * buffer)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  guint payload_len;
+  const guint8 *data = NULL;
+  GstFlowReturn ret;
+  guint available;
+
+  guint maxptime_octets = G_MAXUINT;
+  guint minptime_octets = 0;
+  guint min_payload_len;
+  guint max_payload_len;
+  gboolean use_adapter = FALSE;
+
+  guint fragment_size;
+
+  ret = GST_FLOW_OK;
+
+  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 is in bits and is converted into multiple bytes */
+  fragment_size = basertpaudiopayload->sample_size;
+  while ((fragment_size % 8) != 0)
+    fragment_size += fragment_size;
+  fragment_size /= 8;
+
+  /* max number of bytes based on given ptime */
+  if (basepayload->max_ptime != -1) {
+    maxptime_octets = 8 * basepayload->max_ptime * basepayload->clock_rate /
+        (basertpaudiopayload->sample_size * GST_SECOND);
+  }
+
+  max_payload_len = MIN (
+      /* MTU max */
+      gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+          (basertpaudiopayload), 0, 0),
+      /* ptime max */
+      maxptime_octets);
+
+  /* min number of bytes based on a given ptime, has to be a multiple
+     of sample rate */
+  minptime_octets = 8 * basepayload->min_ptime * basepayload->clock_rate /
+      (basertpaudiopayload->sample_size * GST_SECOND);
+
+  min_payload_len = MAX (minptime_octets, fragment_size);
+
+  if (min_payload_len > max_payload_len) {
+    min_payload_len = max_payload_len;
+  }
+
+  GST_DEBUG_OBJECT (basertpaudiopayload,
+      "Calculated min_payload_len %u and max_payload_len %u",
+      min_payload_len, max_payload_len);
+
+  if (basertpaudiopayload->priv->adapter &&
+      gst_adapter_available (basertpaudiopayload->priv->adapter)) {
+    /* If there is always data in the adapter, we have to use it */
+    gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
+    available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+    use_adapter = TRUE;
+  } else {
+    /* let's set the base timestamp */
+    basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
+
+    /* If buffer fits on an RTP packet, let's just push it through */
+    /* this will check against max_ptime and max_mtu */
+    if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
+        GST_BUFFER_SIZE (buffer) <= max_payload_len) {
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+          GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
+          GST_BUFFER_TIMESTAMP (buffer));
+      gst_buffer_unref (buffer);
+
+      return ret;
+    }
+
+    available = GST_BUFFER_SIZE (buffer);
+    data = (guint8 *) GST_BUFFER_DATA (buffer);
+  }
+
+  while (available >= min_payload_len) {
+    gfloat num, datarate;
+
+    payload_len =
+        MIN (max_payload_len, (available / fragment_size) * fragment_size);
+
+    if (use_adapter) {
+      data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
+    }
+
+    ret =
+        gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
+        basertpaudiopayload->base_ts);
+
+    num = payload_len * 8;
+    datarate = (basertpaudiopayload->sample_size * basepayload->clock_rate);
+
+    basertpaudiopayload->base_ts +=
+        /* payload_len (bits) * nsecs/sec / datarate (bits*sec) */
+        gst_gdouble_to_guint64 (num / datarate * GST_SECOND);
+    GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (basertpaudiopayload->base_ts));
+
+    if (use_adapter) {
+      gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
+      available = gst_adapter_available (basertpaudiopayload->priv->adapter);
+    } else {
+      available -= payload_len;
+      data += payload_len;
+    }
+  }
+
+  if (!use_adapter) {
+    if (available != 0 && basertpaudiopayload->priv->adapter) {
+      GstBuffer *buf;
+
+      buf = gst_buffer_create_sub (buffer,
+          GST_BUFFER_SIZE (buffer) - available, available);
+      gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
+    }
+    gst_buffer_unref (buffer);
+  }
+
+  return ret;
+}
+
+/**
+ * gst_base_rtp_audio_payload_push:
+ * @baseaudiopayload: a #GstBaseRTPPayload
+ * @data: data to set as payload
+ * @payload_len: length of payload
+ * @timestamp: a #GstClockTime
+ *
+ * Create an RTP buffer and store @payload_len bytes of @data as the
+ * payload. Set the timestamp on the new buffer to @timestamp before pushing
+ * the buffer downstream.
+ *
+ * Returns: a #GstFlowReturn
+ *
+ * Since: 0.10.13
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstFlowReturn
+gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
+    const guint8 * data, guint payload_len, GstClockTime timestamp)
+{
+  GstBaseRTPPayload *basepayload;
+  GstBuffer *outbuf;
+  guint8 *payload;
+  GstFlowReturn ret;
+
+  basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+
+  GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
+      payload_len, GST_TIME_ARGS (timestamp));
+
+  /* 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;
+}
+
+static GstStateChangeReturn
+gst_base_rtp_payload_audio_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstBaseRTPAudioPayload *basertppayload;
+  GstStateChangeReturn ret;
+
+  basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element);
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      if (basertppayload->priv->adapter) {
+        gst_adapter_clear (basertppayload->priv->adapter);
+      }
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gboolean res = FALSE;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      if (basertpaudiopayload->priv->adapter) {
+        gst_adapter_clear (basertpaudiopayload->priv->adapter);
+      }
+      break;
+    case GST_EVENT_FLUSH_STOP:
+      if (basertpaudiopayload->priv->adapter) {
+        gst_adapter_clear (basertpaudiopayload->priv->adapter);
+      }
+      break;
+    default:
+      break;
+  }
+
+  gst_object_unref (basertpaudiopayload);
+
+  /* return FALSE to let parent handle the remainder of the event */
+  return res;
+}
+
+/**
+ * gst_base_rtp_audio_payload_get_adapter:
+ * @basertpaudiopayload: a #GstBaseRTPAudioPayload
+ *
+ * Gets the internal adapter used by the depayloader.
+ *
+ * Returns: a #GstAdapter.
+ *
+ * Since: 0.10.13
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstAdapter *
+gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
+    * basertpaudiopayload)
+{
+  GstAdapter *adapter;
+
+  if ((adapter = basertpaudiopayload->priv->adapter))
+    g_object_ref (adapter);
+
+  return adapter;
+}