/* 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:gstrtpbaseaudiopayload

 * @title: GstRTPBaseAudioPayload

 * @short_description: Base class for audio RTP payloader

 *

 * Provides a base class for audio RTP payloaders for frame or sample based

 * audio codecs (constant bitrate)

 *

 * This class derives from GstRTPBasePayload. 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.

 *

 * ## Usage

 *

 * To use this base class, your child element needs to call either

 * gst_rtp_base_audio_payload_set_frame_based() or

 * gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the

 * element's _init() function. Then, the child element must call either

 * gst_rtp_base_audio_payload_set_frame_options(),

 * gst_rtp_base_audio_payload_set_sample_options() or

 * gst_rtp_base_audio_payload_set_samplebits_options. Since

 * GstRTPBaseAudioPayload derives from GstRTPBasePayload, 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 GstRTPBaseAudioPayload.

 *

 */

#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 <gst/audio/audio.h>

#include "gstrtpbaseaudiopayload.h"

GST_DEBUG_CATEGORY_STATIC (rtpbaseaudiopayload_debug);

#define GST_CAT_DEFAULT (rtpbaseaudiopayload_debug)

#define DEFAULT_BUFFER_LIST             FALSE

enum

{

  PROP_0,

  PROP_BUFFER_LIST,

  PROP_LAST

};

/* function to convert bytes to a time */

typedef GstClockTime (*GetBytesToTimeFunc) (GstRTPBaseAudioPayload * payload,

    guint64 bytes);

/* function to convert bytes to a RTP time */

typedef guint32 (*GetBytesToRTPTimeFunc) (GstRTPBaseAudioPayload * payload,

    guint64 bytes);

/* function to convert time to bytes */

typedef guint64 (*GetTimeToBytesFunc) (GstRTPBaseAudioPayload * payload,

    GstClockTime time);

struct _GstRTPBaseAudioPayloadPrivate

{

  GetBytesToTimeFunc bytes_to_time;

  GetBytesToRTPTimeFunc bytes_to_rtptime;

  GetTimeToBytesFunc time_to_bytes;

  GstAdapter *adapter;

  guint fragment_size;

  GstClockTime frame_duration_ns;

  gboolean discont;

  guint64 offset;

  GstClockTime last_timestamp;

  guint32 last_rtptime;

  guint align;

  guint cached_mtu;

  guint cached_min_ptime;

  guint cached_max_ptime;

  guint cached_ptime;

  guint cached_min_length;

  guint cached_max_length;

  guint cached_ptime_multiple;

  guint cached_align;

  gboolean buffer_list;

};

#define GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE(o) \

  (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_BASE_AUDIO_PAYLOAD, \

                                GstRTPBaseAudioPayloadPrivate))

static void gst_rtp_base_audio_payload_finalize (GObject * object);

static void gst_rtp_base_audio_payload_set_property (GObject * object,

    guint prop_id, const GValue * value, GParamSpec * pspec);

static void gst_rtp_base_audio_payload_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec);

/* bytes to time functions */

static GstClockTime

gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *

    payload, guint64 bytes);

static GstClockTime

gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *

    payload, guint64 bytes);

/* bytes to RTP time functions */

static guint32

gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *

    payload, guint64 bytes);

static guint32

gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *

    payload, guint64 bytes);

/* time to bytes functions */

static guint64

gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *

    payload, GstClockTime time);

static guint64

gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *

    payload, GstClockTime time);

static GstFlowReturn gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload

    * payload, GstBuffer * buffer);

static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement

    * element, GstStateChange transition);

static gboolean gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload

    * payload, GstEvent * event);

#define gst_rtp_base_audio_payload_parent_class parent_class

G_DEFINE_TYPE (GstRTPBaseAudioPayload, gst_rtp_base_audio_payload,

    GST_TYPE_RTP_BASE_PAYLOAD);

static void

gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstRTPBasePayloadClass *gstrtpbasepayload_class;

  g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate));

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

  gobject_class->finalize = gst_rtp_base_audio_payload_finalize;

  gobject_class->set_property = gst_rtp_base_audio_payload_set_property;

  gobject_class->get_property = gst_rtp_base_audio_payload_get_property;

  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,

      g_param_spec_boolean ("buffer-list", "Buffer List",

          "Use Buffer Lists",

          DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  gstelement_class->change_state =

      GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state);

  gstrtpbasepayload_class->handle_buffer =

      GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer);

  gstrtpbasepayload_class->sink_event =

      GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_sink_event);

  GST_DEBUG_CATEGORY_INIT (rtpbaseaudiopayload_debug, "rtpbaseaudiopayload", 0,

      "base audio RTP payloader");

}

static void

gst_rtp_base_audio_payload_init (GstRTPBaseAudioPayload * payload)

{

  payload->priv = GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE (payload);

  /* these need to be set by child object if frame based */

  payload->frame_size = 0;

  payload->frame_duration = 0;

  /* these need to be set by child object if sample based */

  payload->sample_size = 0;

  payload->priv->adapter = gst_adapter_new ();

  payload->priv->buffer_list = DEFAULT_BUFFER_LIST;

}

static void

gst_rtp_base_audio_payload_finalize (GObject * object)

{

  GstRTPBaseAudioPayload *payload;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);

  g_object_unref (payload->priv->adapter);

  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));

}

static void

gst_rtp_base_audio_payload_set_property (GObject * object,

    guint prop_id, const GValue * value, GParamSpec * pspec)

{

  GstRTPBaseAudioPayload *payload;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);

  switch (prop_id) {

    case PROP_BUFFER_LIST:

#if 0

      payload->priv->buffer_list = g_value_get_boolean (value);

#endif

      payload->priv->buffer_list = FALSE;

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_rtp_base_audio_payload_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec)

{

  GstRTPBaseAudioPayload *payload;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);

  switch (prop_id) {

    case PROP_BUFFER_LIST:

      g_value_set_boolean (value, payload->priv->buffer_list);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

/**

 * gst_rtp_base_audio_payload_set_frame_based:

 * @rtpbaseaudiopayload: a pointer to the element.

 *

 * Tells #GstRTPBaseAudioPayload that the child element is for a frame based

 * audio codec

 */

void

gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *

    rtpbaseaudiopayload)

{

  g_return_if_fail (rtpbaseaudiopayload != NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);

  rtpbaseaudiopayload->priv->bytes_to_time =

      gst_rtp_base_audio_payload_frame_bytes_to_time;

  rtpbaseaudiopayload->priv->bytes_to_rtptime =

      gst_rtp_base_audio_payload_frame_bytes_to_rtptime;

  rtpbaseaudiopayload->priv->time_to_bytes =

      gst_rtp_base_audio_payload_frame_time_to_bytes;

}

/**

 * gst_rtp_base_audio_payload_set_sample_based:

 * @rtpbaseaudiopayload: a pointer to the element.

 *

 * Tells #GstRTPBaseAudioPayload that the child element is for a sample based

 * audio codec

 */

void

gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *

    rtpbaseaudiopayload)

{

  g_return_if_fail (rtpbaseaudiopayload != NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);

  g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);

  rtpbaseaudiopayload->priv->bytes_to_time =

      gst_rtp_base_audio_payload_sample_bytes_to_time;

  rtpbaseaudiopayload->priv->bytes_to_rtptime =

      gst_rtp_base_audio_payload_sample_bytes_to_rtptime;

  rtpbaseaudiopayload->priv->time_to_bytes =

      gst_rtp_base_audio_payload_sample_time_to_bytes;

}

/**

 * gst_rtp_base_audio_payload_set_frame_options:

 * @rtpbaseaudiopayload: 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.

 *

 */

void

gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload

    * rtpbaseaudiopayload, gint frame_duration, gint frame_size)

{

  GstRTPBaseAudioPayloadPrivate *priv;

  g_return_if_fail (rtpbaseaudiopayload != NULL);

  priv = rtpbaseaudiopayload->priv;

  rtpbaseaudiopayload->frame_duration = frame_duration;

  priv->frame_duration_ns = frame_duration * GST_MSECOND;

  rtpbaseaudiopayload->frame_size = frame_size;

  priv->align = frame_size;

  gst_adapter_clear (priv->adapter);

  GST_DEBUG_OBJECT (rtpbaseaudiopayload, "frame set to %d ms and size %d",

      frame_duration, frame_size);

}

/**

 * gst_rtp_base_audio_payload_set_sample_options:

 * @rtpbaseaudiopayload: a pointer to the element.

 * @sample_size: Size per sample in bytes.

 *

 * Sets the options for sample based audio codecs.

 */

void

gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload

    * rtpbaseaudiopayload, gint sample_size)

{

  g_return_if_fail (rtpbaseaudiopayload != NULL);

  /* sample_size is in bits internally */

  gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,

      sample_size * 8);

}

/**

 * gst_rtp_base_audio_payload_set_samplebits_options:

 * @rtpbaseaudiopayload: a pointer to the element.

 * @sample_size: Size per sample in bits.

 *

 * Sets the options for sample based audio codecs.

 */

void

gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload

    * rtpbaseaudiopayload, gint sample_size)

{

  guint fragment_size;

  GstRTPBaseAudioPayloadPrivate *priv;

  g_return_if_fail (rtpbaseaudiopayload != NULL);

  priv = rtpbaseaudiopayload->priv;

  rtpbaseaudiopayload->sample_size = sample_size;

  /* sample_size is in bits and is converted into multiple bytes */

  fragment_size = sample_size;

  while ((fragment_size % 8) != 0)

    fragment_size += fragment_size;

  priv->fragment_size = fragment_size / 8;

  priv->align = priv->fragment_size;

  gst_adapter_clear (priv->adapter);

  GST_DEBUG_OBJECT (rtpbaseaudiopayload,

      "Samplebits set to sample size %d bits", sample_size);

}

static void

gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload,

    GstBuffer * buffer, guint payload_len, GstClockTime timestamp)

{

  GstRTPBasePayload *basepayload;

  GstRTPBaseAudioPayloadPrivate *priv;

  GstRTPBuffer rtp = { NULL };

  basepayload = GST_RTP_BASE_PAYLOAD_CAST (payload);

  priv = payload->priv;

  /* set payload type */

  gst_rtp_buffer_map (buffer, GST_MAP_WRITE, &rtp;;

  gst_rtp_buffer_set_payload_type (&rtp, basepayload->pt);

  /* set marker bit for disconts */

  if (priv->discont) {

    GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT");

    gst_rtp_buffer_set_marker (&rtp, TRUE);

    GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);

    priv->discont = FALSE;

  }

  gst_rtp_buffer_unmap (&rtp;;

  GST_BUFFER_PTS (buffer) = timestamp;

  /* get the offset in RTP time */

  GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset);

  priv->offset += payload_len;

  /* Set the duration from the size */

  GST_BUFFER_DURATION (buffer) = priv->bytes_to_time (payload, payload_len);

  /* remember the last rtptime/timestamp pair. We will use this to realign our

   * RTP timestamp after a buffer discont */

  priv->last_rtptime = GST_BUFFER_OFFSET (buffer);

  priv->last_timestamp = timestamp;

}

/**

 * gst_rtp_base_audio_payload_push:

 * @baseaudiopayload: a #GstRTPBasePayload

 * @data: (array length=payload_len): 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

 */

GstFlowReturn

gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,

    const guint8 * data, guint payload_len, GstClockTime timestamp)

{

  GstRTPBasePayload *basepayload;

  GstBuffer *outbuf;

  guint8 *payload;

  GstFlowReturn ret;

  GstRTPBuffer rtp = { NULL };

  basepayload = GST_RTP_BASE_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_map (outbuf, GST_MAP_WRITE, &rtp;;

  payload = gst_rtp_buffer_get_payload (&rtp;;

  memcpy (payload, data, payload_len);

  gst_rtp_buffer_unmap (&rtp;;

  /* set metadata */

  gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,

      timestamp);

  ret = gst_rtp_base_payload_push (basepayload, outbuf);

  return ret;

}

typedef struct

{

  GstRTPBaseAudioPayload *pay;

  GstBuffer *outbuf;

} CopyMetaData;

static gboolean

foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)

{

  CopyMetaData *data = user_data;

  GstRTPBaseAudioPayload *pay = data->pay;

  GstBuffer *outbuf = data->outbuf;

  const GstMetaInfo *info = (*meta)->info;

  const gchar *const *tags = gst_meta_api_type_get_tags (info->api);

  if (info->transform_func && (!tags || (g_strv_length ((gchar **) tags) == 1

              && gst_meta_api_type_has_tag (info->api,

                  g_quark_from_string (GST_META_TAG_AUDIO_STR))))) {

    GstMetaTransformCopy copy_data = { FALSE, 0, -1 };

    GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api));

    /* simply copy then */

    info->transform_func (outbuf, *meta, inbuf,

        _gst_meta_transform_copy, &copy_data);

  } else {

    GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api));

  }

  return TRUE;

}

static GstFlowReturn

gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *

    baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)

{

  GstRTPBasePayload *basepayload;

  GstRTPBaseAudioPayloadPrivate *priv;

  GstBuffer *outbuf;

  guint payload_len;

  GstFlowReturn ret;

  priv = baseaudiopayload->priv;

  basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);

  payload_len = gst_buffer_get_size (buffer);

  GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,

      payload_len, GST_TIME_ARGS (timestamp));

  /* create just the RTP header buffer */

  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);

  /* set metadata */

  gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,

      timestamp);

  if (priv->buffer_list) {

    GstBufferList *list;

    guint i, len;

    list = gst_buffer_list_new ();

    len = gst_buffer_list_length (list);

    for (i = 0; i < len; i++) {

      /* FIXME */

      g_warning ("bufferlist not implemented");

      gst_buffer_list_add (list, outbuf);

      gst_buffer_list_add (list, buffer);

    }

    GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);

    ret = gst_rtp_base_payload_push_list (basepayload, list);

  } else {

    CopyMetaData data;

    /* copy payload */

    data.pay = baseaudiopayload;

    data.outbuf = outbuf;

    gst_buffer_foreach_meta (buffer, foreach_metadata, &data);

    outbuf = gst_buffer_append (outbuf, buffer);

    GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);

    ret = gst_rtp_base_payload_push (basepayload, outbuf);

  }

  return ret;

}

/**

 * gst_rtp_base_audio_payload_flush:

 * @baseaudiopayload: a #GstRTPBasePayload

 * @payload_len: length of payload

 * @timestamp: a #GstClockTime

 *

 * Create an RTP buffer and store @payload_len bytes of the adapter as the

 * payload. Set the timestamp on the new buffer to @timestamp before pushing

 * the buffer downstream.

 *

 * If @payload_len is -1, all pending bytes will be flushed. If @timestamp is

 * -1, the timestamp will be calculated automatically.

 *

 * Returns: a #GstFlowReturn

 */

GstFlowReturn

gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,

    guint payload_len, GstClockTime timestamp)

{

  GstRTPBasePayload *basepayload;

  GstRTPBaseAudioPayloadPrivate *priv;

  GstBuffer *outbuf;

  GstFlowReturn ret;

  GstAdapter *adapter;

  guint64 distance;

  priv = baseaudiopayload->priv;

  adapter = priv->adapter;

  basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);

  if (payload_len == -1)

    payload_len = gst_adapter_available (adapter);

  /* nothing to do, just return */

  if (payload_len == 0)

    return GST_FLOW_OK;

  if (timestamp == -1) {

    /* calculate the timestamp */

    timestamp = gst_adapter_prev_pts (adapter, &distance);

    GST_LOG_OBJECT (baseaudiopayload,

        "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,

        GST_TIME_ARGS (timestamp), distance);

    if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {

      /* convert the number of bytes since the last timestamp to time and add to

       * the last seen timestamp */

      timestamp += priv->bytes_to_time (baseaudiopayload, distance);

    }

  }

  GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,

      payload_len, GST_TIME_ARGS (timestamp));

  if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {

    GstBuffer *buffer;

    /* we can quickly take a buffer out of the adapter without having to copy

     * anything. */

    buffer = gst_adapter_take_buffer (adapter, payload_len);

    ret =

        gst_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,

        timestamp);

  } else {

    GstBuffer *paybuf;

    CopyMetaData data;

    /* create buffer to hold the payload */

    outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);

    paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);

    data.pay = baseaudiopayload;

    data.outbuf = outbuf;

    gst_buffer_foreach_meta (paybuf, foreach_metadata, &data);

    outbuf = gst_buffer_append (outbuf, paybuf);

    /* set metadata */

    gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,

        timestamp);

    ret = gst_rtp_base_payload_push (basepayload, outbuf);

  }

  return ret;

}

#define ALIGN_DOWN(val,len) ((val) - ((val) % (len)))

/* calculate the min and max length of a packet. This depends on the configured

 * mtu and min/max_ptime values. We cache those so that we don't have to redo

 * all the calculations */

static gboolean

gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *

    basepayload, guint * min_payload_len, guint * max_payload_len,

    guint * align)

{

  GstRTPBaseAudioPayload *payload;

  GstRTPBaseAudioPayloadPrivate *priv;

  guint max_mtu, mtu;

  guint maxptime_octets;

  guint minptime_octets;

  guint ptime_mult_octets;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);

  priv = payload->priv;

  if (priv->align == 0)

    return FALSE;

  mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);

  /* check cached values */

  if (G_LIKELY (priv->cached_mtu == mtu

          && priv->cached_ptime_multiple ==

          basepayload->ptime_multiple

          && priv->cached_ptime == basepayload->ptime

          && priv->cached_max_ptime == basepayload->max_ptime

          && priv->cached_min_ptime == basepayload->min_ptime)) {

    /* if nothing changed, return cached values */

    *min_payload_len = priv->cached_min_length;

    *max_payload_len = priv->cached_max_length;

    *align = priv->cached_align;

    return TRUE;

  }

  ptime_mult_octets = priv->time_to_bytes (payload,

      basepayload->ptime_multiple);

  *align = ALIGN_DOWN (MAX (priv->align, ptime_mult_octets), priv->align);

  /* ptime max */

  if (basepayload->max_ptime != -1) {

    maxptime_octets = priv->time_to_bytes (payload, basepayload->max_ptime);

  } else {

    maxptime_octets = G_MAXUINT;

  }

  /* MTU max */

  max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);

  /* round down to alignment */

  max_mtu = ALIGN_DOWN (max_mtu, *align);

  /* combine max ptime and max payload length */

  *max_payload_len = MIN (max_mtu, maxptime_octets);

  /* min number of bytes based on a given ptime */

  minptime_octets = priv->time_to_bytes (payload, basepayload->min_ptime);

  /* must be at least one frame size */

  *min_payload_len = MAX (minptime_octets, *align);

  if (*min_payload_len > *max_payload_len)

    *min_payload_len = *max_payload_len;

  /* If the ptime is specified in the caps, tried to adhere to it exactly */

  if (basepayload->ptime) {

    guint ptime_in_bytes = priv->time_to_bytes (payload,

        basepayload->ptime);

    /* clip to computed min and max lengths */

    ptime_in_bytes = MAX (*min_payload_len, ptime_in_bytes);

    ptime_in_bytes = MIN (*max_payload_len, ptime_in_bytes);

    *min_payload_len = *max_payload_len = ptime_in_bytes;

  }

  /* cache values */

  priv->cached_mtu = mtu;

  priv->cached_ptime = basepayload->ptime;

  priv->cached_min_ptime = basepayload->min_ptime;

  priv->cached_max_ptime = basepayload->max_ptime;

  priv->cached_ptime_multiple = basepayload->ptime_multiple;

  priv->cached_min_length = *min_payload_len;

  priv->cached_max_length = *max_payload_len;

  priv->cached_align = *align;

  return TRUE;

}

/* frame conversions functions */

static GstClockTime

gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *

    payload, guint64 bytes)

{

  guint64 framecount;

  framecount = bytes / payload->frame_size;

  if (G_UNLIKELY (bytes % payload->frame_size))

    framecount++;

  return framecount * payload->priv->frame_duration_ns;

}

static guint32

gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *

    payload, guint64 bytes)

{

  guint64 framecount;

  guint64 time;

  framecount = bytes / payload->frame_size;

  if (G_UNLIKELY (bytes % payload->frame_size))

    framecount++;

  time = framecount * payload->priv->frame_duration_ns;

  return gst_util_uint64_scale_int (time,

      GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);

}

static guint64

gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *

    payload, GstClockTime time)

{

  return gst_util_uint64_scale (time, payload->frame_size,

      payload->priv->frame_duration_ns);

}

/* sample conversion functions */

static GstClockTime

gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *

    payload, guint64 bytes)

{

  guint64 rtptime;

  /* avoid division when we can */

  if (G_LIKELY (payload->sample_size != 8))

    rtptime = gst_util_uint64_scale_int (bytes, 8, payload->sample_size);

  else

    rtptime = bytes;

  return gst_util_uint64_scale_int (rtptime, GST_SECOND,

      GST_RTP_BASE_PAYLOAD (payload)->clock_rate);

}

static guint32

gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *

    payload, guint64 bytes)

{

  /* avoid division when we can */

  if (G_LIKELY (payload->sample_size != 8))

    return gst_util_uint64_scale_int (bytes, 8, payload->sample_size);

  else

    return bytes;

}

static guint64

gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *

    payload, guint64 time)

{

  guint64 samples;

  samples = gst_util_uint64_scale_int (time,

      GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);

  /* avoid multiplication when we can */

  if (G_LIKELY (payload->sample_size != 8))

    return gst_util_uint64_scale_int (samples, payload->sample_size, 8);

  else

    return samples;

}

static GstFlowReturn

gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *

    basepayload, GstBuffer * buffer)

{

  GstRTPBaseAudioPayload *payload;

  GstRTPBaseAudioPayloadPrivate *priv;

  guint payload_len;

  GstFlowReturn ret;

  guint available;

  guint min_payload_len;

  guint max_payload_len;

  guint align;

  guint size;

  gboolean discont;

  GstClockTime timestamp;

  ret = GST_FLOW_OK;

  payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);

  priv = payload->priv;

  timestamp = GST_BUFFER_PTS (buffer);

  discont = GST_BUFFER_IS_DISCONT (buffer);

  if (discont) {

    GST_DEBUG_OBJECT (payload, "Got DISCONT");

    /* flush everything out of the adapter, mark DISCONT */

    ret = gst_rtp_base_audio_payload_flush (payload, -1, -1);

    priv->discont = TRUE;

    /* get the distance between the timestamp gap and produce the same gap in

     * the RTP timestamps */

    if (priv->last_timestamp != -1 && timestamp != -1) {

      /* we had a last timestamp, compare it to the new timestamp and update the

       * offset counter for RTP timestamps. The effect is that we will produce

       * output buffers containing the same RTP timestamp gap as the gap

       * between the GST timestamps. */

      if (timestamp > priv->last_timestamp) {

        GstClockTime diff;

        guint64 bytes;

        /* we're only going to apply a positive gap, otherwise we let the marker

         * bit do its thing. simply convert to bytes and add the current

         * offset */

        diff = timestamp - priv->last_timestamp;

        bytes = priv->time_to_bytes (payload, diff);

        priv->offset += bytes;

        GST_DEBUG_OBJECT (payload,

            "elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT

            ", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,