/* GStreamer

 * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>

 *

 * 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.

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include <string.h>

#include <stdlib.h>

#include <gst/rtp/gstrtpbuffer.h>

#include "gstrtpmp4gdepay.h"

#include "gstrtputils.h"

GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug);

#define GST_CAT_DEFAULT (rtpmp4gdepay_debug)

static GstStaticPadTemplate gst_rtp_mp4g_depay_src_template =

    GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("video/mpeg,"

        "mpegversion=(int) 4,"

        "systemstream=(boolean)false;"

        "audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string)raw")

    );

static GstStaticPadTemplate gst_rtp_mp4g_depay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("application/x-rtp, "

        "media = (string) { \"video\", \"audio\", \"application\" }, "

        "clock-rate = (int) [1, MAX ], "

        "encoding-name = (string) \"MPEG4-GENERIC\", "

        /* required string params */

        /* "streamtype = (string) { \"4\", \"5\" }, "  Not set by Wowza    4 = video, 5 = audio */

        /* "profile-level-id = (string) [1,MAX], " */

        /* "config = (string) [1,MAX]" */

        "mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" } "

        /* Optional general parameters */

        /* "objecttype = (string) [1,MAX], " */

        /* "constantsize = (string) [1,MAX], " *//* constant size of each AU */

        /* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */

        /* "maxdisplacement = (string) [1,MAX], " */

        /* "de-interleavebuffersize = (string) [1,MAX], " */

        /* Optional configuration parameters */

        /* "sizelength = (string) [1, 32], " */

        /* "indexlength = (string) [1, 32], " */

        /* "indexdeltalength = (string) [1, 32], " */

        /* "ctsdeltalength = (string) [1, 32], " */

        /* "dtsdeltalength = (string) [1, 32], " */

        /* "randomaccessindication = (string) {0, 1}, " */

        /* "streamstateindication = (string) [0, 32], " */

        /* "auxiliarydatasizelength = (string) [0, 32]" */ )

    );

/* simple bitstream parser */

typedef struct

{

  const guint8 *data;

  const guint8 *end;

  gint head;                    /* bitpos in the cache of next bit */

  guint64 cache;                /* cached bytes */

} GstBsParse;

static void

gst_bs_parse_init (GstBsParse * bs, const guint8 * data, guint size)

{

  bs->data = data;

  bs->end = data + size;

  bs->head = 0;

  bs->cache = 0xffffffff;

}

static guint32

gst_bs_parse_read (GstBsParse * bs, guint n)

{

  guint32 res = 0;

  gint shift;

  if (n == 0)

    return res;

  /* fill up the cache if we need to */

  while (bs->head < n) {

    if (bs->data >= bs->end) {

      /* we're at the end, can't produce more than head number of bits */

      n = bs->head;

      break;

    }

    /* shift bytes in cache, moving the head bits of the cache left */

    bs->cache = (bs->cache << 8) | *bs->data++;

    bs->head += 8;

  }

  /* bring the required bits down and truncate */

  if ((shift = bs->head - n) > 0)

    res = bs->cache >> shift;

  else

    res = bs->cache;

  /* mask out required bits */

  if (n < 32)

    res &= (1 << n) - 1;

  bs->head = shift;

  return res;

}

#define gst_rtp_mp4g_depay_parent_class parent_class

G_DEFINE_TYPE (GstRtpMP4GDepay, gst_rtp_mp4g_depay,

    GST_TYPE_RTP_BASE_DEPAYLOAD);

static void gst_rtp_mp4g_depay_finalize (GObject * object);

static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload,

    GstCaps * caps);

static GstBuffer *gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload,

    GstRTPBuffer * rtp);

static gboolean gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter,

    GstEvent * event);

static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement *

    element, GstStateChange transition);

static void

gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;

  gobject_class->finalize = gst_rtp_mp4g_depay_finalize;

  gstelement_class->change_state = gst_rtp_mp4g_depay_change_state;

  gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4g_depay_process;

  gstrtpbasedepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps;

  gstrtpbasedepayload_class->handle_event = gst_rtp_mp4g_depay_handle_event;

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_mp4g_depay_src_template);

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_mp4g_depay_sink_template);

  gst_element_class_set_static_metadata (gstelement_class,

      "RTP MPEG4 ES depayloader", "Codec/Depayloader/Network/RTP",

      "Extracts MPEG4 elementary streams from RTP packets (RFC 3640)",

      "Wim Taymans <wim.taymans@gmail.com>");

  GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0,

      "MP4-generic RTP Depayloader");

}

static void

gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay)

{

  rtpmp4gdepay->adapter = gst_adapter_new ();

  rtpmp4gdepay->packets = g_queue_new ();

}

static void

gst_rtp_mp4g_depay_finalize (GObject * object)

{

  GstRtpMP4GDepay *rtpmp4gdepay;

  rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object);

  g_object_unref (rtpmp4gdepay->adapter);

  rtpmp4gdepay->adapter = NULL;

  g_queue_free (rtpmp4gdepay->packets);

  rtpmp4gdepay->packets = NULL;

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static gint

gst_rtp_mp4g_depay_parse_int (GstStructure * structure, const gchar * field,

    gint def)

{

  const gchar *str;

  gint res;

  if ((str = gst_structure_get_string (structure, field)))

    return atoi (str);

  if (gst_structure_get_int (structure, field, &res))

    return res;

  return def;

}

static gboolean

gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)

{

  GstStructure *structure;

  GstRtpMP4GDepay *rtpmp4gdepay;

  GstCaps *srccaps = NULL;

  const gchar *str;

  gint clock_rate;

  gint someint;

  gboolean res;

  rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))

    clock_rate = 90000;         /* default */

  depayload->clock_rate = clock_rate;

  if ((str = gst_structure_get_string (structure, "media"))) {

    if (strcmp (str, "audio") == 0) {

      srccaps = gst_caps_new_simple ("audio/mpeg",

          "mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw",

          NULL);

    } else if (strcmp (str, "video") == 0) {

      srccaps = gst_caps_new_simple ("video/mpeg",

          "mpegversion", G_TYPE_INT, 4,

          "systemstream", G_TYPE_BOOLEAN, FALSE, NULL);

    }

  }

  if (srccaps == NULL)

    goto unknown_media;

  /* these values are optional and have a default value of 0 (no header) */

  rtpmp4gdepay->sizelength =

      gst_rtp_mp4g_depay_parse_int (structure, "sizelength", 0);

  rtpmp4gdepay->indexlength =

      gst_rtp_mp4g_depay_parse_int (structure, "indexlength", 0);

  rtpmp4gdepay->indexdeltalength =

      gst_rtp_mp4g_depay_parse_int (structure, "indexdeltalength", 0);

  rtpmp4gdepay->ctsdeltalength =

      gst_rtp_mp4g_depay_parse_int (structure, "ctsdeltalength", 0);

  rtpmp4gdepay->dtsdeltalength =

      gst_rtp_mp4g_depay_parse_int (structure, "dtsdeltalength", 0);

  someint =

      gst_rtp_mp4g_depay_parse_int (structure, "randomaccessindication", 0);

  rtpmp4gdepay->randomaccessindication = someint > 0 ? 1 : 0;

  rtpmp4gdepay->streamstateindication =

      gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0);

  rtpmp4gdepay->auxiliarydatasizelength =

      gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0);

  rtpmp4gdepay->constantSize =

      gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0);

  rtpmp4gdepay->constantDuration =

      gst_rtp_mp4g_depay_parse_int (structure, "constantduration", 0);

  rtpmp4gdepay->maxDisplacement =

      gst_rtp_mp4g_depay_parse_int (structure, "maxdisplacement", 0);

  /* get config string */

  if ((str = gst_structure_get_string (structure, "config"))) {

    GValue v = { 0 };

    g_value_init (&v, GST_TYPE_BUFFER);

    if (gst_value_deserialize (&v, str)) {

      GstBuffer *buffer;

      buffer = gst_value_get_buffer (&v);

      gst_caps_set_simple (srccaps,

          "codec_data", GST_TYPE_BUFFER, buffer, NULL);

      g_value_unset (&v);

    } else {

      g_warning ("cannot convert config to buffer");

    }

  }

  res = gst_pad_set_caps (depayload->srcpad, srccaps);

  gst_caps_unref (srccaps);

  return res;

  /* ERRORS */

unknown_media:

  {

    GST_DEBUG_OBJECT (rtpmp4gdepay, "Unknown media type");

    return FALSE;

  }

}

static void

gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay)

{

  GstBuffer *outbuf;

  while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets)))

    gst_buffer_unref (outbuf);

}

static void

gst_rtp_mp4g_depay_reset (GstRtpMP4GDepay * rtpmp4gdepay)

{

  gst_adapter_clear (rtpmp4gdepay->adapter);

  rtpmp4gdepay->max_AU_index = -1;

  rtpmp4gdepay->next_AU_index = -1;

  rtpmp4gdepay->prev_AU_index = -1;

  rtpmp4gdepay->prev_rtptime = -1;

  rtpmp4gdepay->last_AU_index = -1;

  gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay);

}

static void

gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)

{

  GstBuffer *outbuf;

  gboolean discont = FALSE;

  guint AU_index;

  while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {

    AU_index = GST_BUFFER_OFFSET (outbuf);

    GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);

    if (rtpmp4gdepay->next_AU_index != AU_index) {

      GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",

          rtpmp4gdepay->next_AU_index);

      discont = TRUE;

    }

    if (discont) {

      GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);

      discont = FALSE;

    }

    GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing AU_index %u", AU_index);

    gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0);

    gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);

    rtpmp4gdepay->next_AU_index = AU_index + 1;

  }

}

static void

gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf)

{

  guint AU_index = GST_BUFFER_OFFSET (outbuf);

  if (rtpmp4gdepay->next_AU_index == -1) {

    GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index);

    rtpmp4gdepay->next_AU_index = AU_index;

  }

  if (rtpmp4gdepay->next_AU_index == AU_index) {

    GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index);

    /* we received the expected packet, push it and flush as much as we can from

     * the queue */

    gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0);

    gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);

    rtpmp4gdepay->next_AU_index++;

    while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) {

      AU_index = GST_BUFFER_OFFSET (outbuf);

      GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);

      if (rtpmp4gdepay->next_AU_index == AU_index) {

        GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u",

            AU_index);

        outbuf = g_queue_pop_head (rtpmp4gdepay->packets);

        gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0);

        gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay),

            outbuf);

        rtpmp4gdepay->next_AU_index++;

      } else {

        GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u",

            rtpmp4gdepay->next_AU_index);

        break;

      }

    }

  } else {

    GList *list;

    GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index);

    /* loop the list to skip strictly smaller AU_index buffers */

    for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) {

      guint idx;

      gint gap;

      idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data));

      /* compare the new seqnum to the one in the buffer */

      gap = (gint) (idx - AU_index);

      GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx,

          gap);

      /* AU_index <= idx, we can stop looking */

      if (G_LIKELY (gap > 0))

        break;

    }

    if (G_LIKELY (list))

      g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf);

    else

      g_queue_push_tail (rtpmp4gdepay->packets, outbuf);

  }

}

static GstBuffer *

gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)

{

  GstRtpMP4GDepay *rtpmp4gdepay;

  GstBuffer *outbuf = NULL;

  GstClockTime timestamp;

  rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);

  /* flush remaining data on discont */

  if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {

    GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");

    gst_adapter_clear (rtpmp4gdepay->adapter);

  }

  timestamp = GST_BUFFER_PTS (rtp->buffer);

  {

    gint payload_len, payload_AU;

    guint8 *payload;

    guint32 rtptime;

    guint AU_headers_len;

    guint AU_size, AU_index, AU_index_delta, payload_AU_size;

    gboolean M;

    payload_len = gst_rtp_buffer_get_payload_len (rtp);

    payload = gst_rtp_buffer_get_payload (rtp);

    GST_DEBUG_OBJECT (rtpmp4gdepay, "received payload of %d", payload_len);

    rtptime = gst_rtp_buffer_get_timestamp (rtp);

    M = gst_rtp_buffer_get_marker (rtp);

    if (rtpmp4gdepay->sizelength > 0) {

      gint num_AU_headers, AU_headers_bytes, i;

      GstBsParse bs;

      if (payload_len < 2)

        goto short_payload;

      /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+

       * |AU-headers-length|AU-header|AU-header|      |AU-header|padding|

       * |                 |   (1)   |   (2)   |      |   (n) * | bits  |

       * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+

       *

       * The length is 2 bytes and contains the length of the following

       * AU-headers in bits.

       */

      AU_headers_len = (payload[0] << 8) | payload[1];

      AU_headers_bytes = (AU_headers_len + 7) / 8;

      num_AU_headers = AU_headers_len / 16;

      GST_DEBUG_OBJECT (rtpmp4gdepay, "AU headers len %d, bytes %d, num %d",

          AU_headers_len, AU_headers_bytes, num_AU_headers);

      /* skip header */

      payload += 2;

      payload_len -= 2;

      if (payload_len < AU_headers_bytes)

        goto short_payload;

      /* skip special headers, point to first payload AU */

      payload_AU = 2 + AU_headers_bytes;

      payload_AU_size = payload_len - AU_headers_bytes;

      if (G_UNLIKELY (rtpmp4gdepay->auxiliarydatasizelength)) {

        gint aux_size;

        /* point the bitstream parser to the first auxiliary data bit */

        gst_bs_parse_init (&bs, payload + AU_headers_bytes,

            payload_len - AU_headers_bytes);

        aux_size =

            gst_bs_parse_read (&bs, rtpmp4gdepay->auxiliarydatasizelength);

        /* convert to bytes */

        aux_size = (aux_size + 7) / 8;

        /* AU data then follows auxiliary data */

        if (payload_AU_size < aux_size)

          goto short_payload;

        payload_AU += aux_size;

        payload_AU_size -= aux_size;

      }

      /* point the bitstream parser to the first AU header bit */

      gst_bs_parse_init (&bs, payload, payload_len);

      AU_index = AU_index_delta = 0;

      for (i = 0; i < num_AU_headers && payload_AU_size > 0; i++) {

        /* parse AU header

         *  +---------------------------------------+

         *  |     AU-size                           |

         *  +---------------------------------------+

         *  |     AU-Index / AU-Index-delta         |

         *  +---------------------------------------+

         *  |     CTS-flag                          |

         *  +---------------------------------------+

         *  |     CTS-delta                         |

         *  +---------------------------------------+

         *  |     DTS-flag                          |

         *  +---------------------------------------+

         *  |     DTS-delta                         |

         *  +---------------------------------------+

         *  |     RAP-flag                          |

         *  +---------------------------------------+

         *  |     Stream-state                      |

         *  +---------------------------------------+

         */

        AU_size = gst_bs_parse_read (&bs, rtpmp4gdepay->sizelength);

        /* calculate the AU_index, which is only on the first AU of the packet

         * and the AU_index_delta on the other AUs. This will be used to

         * reconstruct the AU ordering when interleaving. */

        if (i == 0) {

          AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexlength);

          GST_DEBUG_OBJECT (rtpmp4gdepay, "AU index %u", AU_index);

          if (AU_index == 0 && rtpmp4gdepay->prev_AU_index == 0) {

            gint diff;

            gint cd;

            /* if we see two consecutive packets with AU_index of 0, we can

             * assume we have constantDuration packets. Since we don't have

             * the index we must use the AU duration to calculate the

             * index. Get the diff between the timestamps first, this can be

             * positive or negative. */

            if (rtpmp4gdepay->prev_rtptime <= rtptime)

              diff = rtptime - rtpmp4gdepay->prev_rtptime;

            else

              diff = -(rtpmp4gdepay->prev_rtptime - rtptime);

            /* if no constantDuration was given, make one */

            if (rtpmp4gdepay->constantDuration != 0) {

              cd = rtpmp4gdepay->constantDuration;

              GST_DEBUG_OBJECT (depayload, "using constantDuration %d", cd);

            } else if (rtpmp4gdepay->prev_AU_num > 0) {

              /* use number of packets and of previous frame */

              cd = diff / rtpmp4gdepay->prev_AU_num;

              GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd);

              if (!GST_BUFFER_IS_DISCONT (rtp->buffer)) {

                /* rfc3640 - 3.2.3.2

                 * if we see two consecutive packets with AU_index of 0 and

                 * there has been no discontinuity, we must conclude that this

                 * value of constantDuration is correct from now on. */

                GST_DEBUG_OBJECT (depayload,

                    "constantDuration of %d detected", cd);

                rtpmp4gdepay->constantDuration = cd;

              }

            } else {

              /* assume this frame has the same number of packets as the

               * previous one */

              cd = diff / num_AU_headers;

              GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd);

            }

            if (cd > 0) {

              /* get the number of packets by dividing with the duration */

              diff /= cd;

            } else {

              diff = 0;

            }

            rtpmp4gdepay->last_AU_index += diff;

            rtpmp4gdepay->prev_AU_index = AU_index;

            AU_index = rtpmp4gdepay->last_AU_index;

            GST_DEBUG_OBJECT (rtpmp4gdepay, "diff %d, AU index %u", diff,

                AU_index);

          } else {

            rtpmp4gdepay->prev_AU_index = AU_index;

            rtpmp4gdepay->last_AU_index = AU_index;

          }

          /* keep track of the higest AU_index */

          if (rtpmp4gdepay->max_AU_index != -1

              && rtpmp4gdepay->max_AU_index <= AU_index) {

            GST_DEBUG_OBJECT (rtpmp4gdepay, "new interleave group, flushing");

            /* a new interleave group started, flush */

            gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);

          }

          if (G_UNLIKELY (!rtpmp4gdepay->maxDisplacement &&

                  rtpmp4gdepay->max_AU_index != -1

                  && rtpmp4gdepay->max_AU_index >= AU_index)) {

            GstBuffer *outbuf;

            /* some broken non-interleaved streams have AU-index jumping around

             * all over the place, apparently assuming receiver disregards */

            GST_DEBUG_OBJECT (rtpmp4gdepay, "non-interleaved broken AU indices;"

                " forcing continuous flush");

            /* reset AU to avoid repeated DISCONT in such case */

            outbuf = g_queue_peek_head (rtpmp4gdepay->packets);

            if (G_LIKELY (outbuf)) {

              rtpmp4gdepay->next_AU_index = GST_BUFFER_OFFSET (outbuf);

              gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);

            }

            /* rebase next_AU_index to current rtp's first AU_index */

            rtpmp4gdepay->next_AU_index = AU_index;

          }

          rtpmp4gdepay->prev_rtptime = rtptime;

          rtpmp4gdepay->prev_AU_num = num_AU_headers;

        } else {

          AU_index_delta =

              gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);

          AU_index += AU_index_delta + 1;

        }

        /* keep track of highest AU_index */

        if (rtpmp4gdepay->max_AU_index == -1

            || AU_index > rtpmp4gdepay->max_AU_index)

          rtpmp4gdepay->max_AU_index = AU_index;

        /* the presentation time offset, a 2s-complement value, we need this to

         * calculate the timestamp on the output packet. */

        if (rtpmp4gdepay->ctsdeltalength > 0) {

          if (gst_bs_parse_read (&bs, 1))

            gst_bs_parse_read (&bs, rtpmp4gdepay->ctsdeltalength);

        }

        /* the decoding time offset, a 2s-complement value */

        if (rtpmp4gdepay->dtsdeltalength > 0) {

          if (gst_bs_parse_read (&bs, 1))

            gst_bs_parse_read (&bs, rtpmp4gdepay->dtsdeltalength);

        }

        /* RAP-flag to indicate that the AU contains a keyframe */

        if (rtpmp4gdepay->randomaccessindication)

          gst_bs_parse_read (&bs, 1);

        /* stream-state */

        if (rtpmp4gdepay->streamstateindication > 0)

          gst_bs_parse_read (&bs, rtpmp4gdepay->streamstateindication);

        GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d, delta %d", AU_size,

            AU_index, AU_index_delta);

        /* fragmented pakets have the AU_size set to the size of the

         * unfragmented AU. */

        if (AU_size > payload_AU_size)

          AU_size = payload_AU_size;

        /* collect stuff in the adapter, strip header from payload and push in

         * the adapter */

        outbuf =

            gst_rtp_buffer_get_payload_subbuffer (rtp, payload_AU, AU_size);

        gst_adapter_push (rtpmp4gdepay->adapter, outbuf);

        if (M) {

          guint avail;

          /* packet is complete, flush */

          avail = gst_adapter_available (rtpmp4gdepay->adapter);

          outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);

          /* copy some of the fields we calculated above on the buffer. We also

           * copy the AU_index so that we can sort the packets in our queue. */

          GST_BUFFER_PTS (outbuf) = timestamp;

          GST_BUFFER_OFFSET (outbuf) = AU_index;

          if (rtpmp4gdepay->constantDuration != 0) {

            /* if we have constantDuration, calculate timestamp for next AU

             * in this RTP packet. */

            timestamp += (rtpmp4gdepay->constantDuration * GST_SECOND) /

                depayload->clock_rate;

          } else {

            /* otherwise, make sure we don't use the timestamp again for other

             * AUs. */

            timestamp = GST_CLOCK_TIME_NONE;

          }

          GST_DEBUG_OBJECT (depayload,

              "pushing buffer of size %" G_GSIZE_FORMAT,

              gst_buffer_get_size (outbuf));

          gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf);

        }

        payload_AU += AU_size;

        payload_AU_size -= AU_size;

      }

    } else {

      /* push complete buffer in adapter */

      outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 0, payload_len);

      gst_adapter_push (rtpmp4gdepay->adapter, outbuf);

      /* if this was the last packet of the VOP, create and push a buffer */

      if (M) {

        guint avail;

        avail = gst_adapter_available (rtpmp4gdepay->adapter);

        outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);

        GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %"

            G_GSIZE_FORMAT, gst_buffer_get_size (outbuf));

        return outbuf;

      }

    }

  }

  return NULL;

  /* ERRORS */

short_payload:

  {

    GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,

        ("Packet payload was too short."), (NULL));

    return NULL;

  }

}

static gboolean

gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter, GstEvent * event)

{

  gboolean ret;

  GstRtpMP4GDepay *rtpmp4gdepay;

  rtpmp4gdepay = GST_RTP_MP4G_DEPAY (filter);

  switch (GST_EVENT_TYPE (event)) {

    case GST_EVENT_FLUSH_STOP:

      gst_rtp_mp4g_depay_reset (rtpmp4gdepay);

      break;

    default:

      break;

  }

  ret =

      GST_RTP_BASE_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event);

  return ret;

}

static GstStateChangeReturn

gst_rtp_mp4g_depay_change_state (GstElement * element,

    GstStateChange transition)

{

  GstRtpMP4GDepay *rtpmp4gdepay;

  GstStateChangeReturn ret;

  rtpmp4gdepay = GST_RTP_MP4G_DEPAY (element);

  switch (transition) {

    case GST_STATE_CHANGE_READY_TO_PAUSED:

      gst_rtp_mp4g_depay_reset (rtpmp4gdepay);

      break;

    default:

      break;

  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  switch (transition) {

    case GST_STATE_CHANGE_PAUSED_TO_READY:

      gst_rtp_mp4g_depay_reset (rtpmp4gdepay);

      break;

    default:

      break;

  }

  return ret;

}

gboolean

gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtpmp4gdepay",

      GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY);

}