/* GStreamer
 * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
 * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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.
 */

/**
 * SECTION:gstrtpbasedepayload
 * @title: GstRTPBaseDepayload
 * @short_description: Base class for RTP depayloader
 *
 * Provides a base class for RTP depayloaders
 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include "gstrtpbasedepayload.h"
#include "gstrtpmeta.h"

GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)

struct _GstRTPBaseDepayloadPrivate
{
  GstClockTime npt_start;
  GstClockTime npt_stop;
  gdouble play_speed;
  gdouble play_scale;
  guint clock_base;

  gboolean discont;
  GstClockTime pts;
  GstClockTime dts;
  GstClockTime duration;

  guint32 last_ssrc;
  guint32 last_seqnum;
  guint32 last_rtptime;
  guint32 next_seqnum;

  gboolean negotiated;

  GstCaps *last_caps;
  GstEvent *segment_event;
  guint32 segment_seqnum;       /* Note: this is a GstEvent seqnum */

  gboolean source_info;
  GstBuffer *input_buffer;
};

/* Filter signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

#define DEFAULT_SOURCE_INFO FALSE

enum
{
  PROP_0,
  PROP_STATS,
  PROP_SOURCE_INFO,
  PROP_LAST
};

static void gst_rtp_base_depayload_finalize (GObject * object);
static void gst_rtp_base_depayload_set_property (GObject * object,
    guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_rtp_base_depayload_get_property (GObject * object,
    guint prop_id, GValue * value, GParamSpec * pspec);

static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
    GstObject * parent, GstBuffer * in);
static GstFlowReturn gst_rtp_base_depayload_chain_list (GstPad * pad,
    GstObject * parent, GstBufferList * list);
static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
    GstObject * parent, GstEvent * event);

static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
    element, GstStateChange transition);

static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
    filter, GstEvent * event);
static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
    filter, GstEvent * event);

static GstElementClass *parent_class = NULL;
static gint private_offset = 0;

static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
    klass);
static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
    GstRTPBaseDepayloadClass * klass);
static GstEvent *create_segment_event (GstRTPBaseDepayload * filter,
    guint rtptime, GstClockTime position);

GType
gst_rtp_base_depayload_get_type (void)
{
  static GType rtp_base_depayload_type = 0;

  if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
    static const GTypeInfo rtp_base_depayload_info = {
      sizeof (GstRTPBaseDepayloadClass),
      NULL,
      NULL,
      (GClassInitFunc) gst_rtp_base_depayload_class_init,
      NULL,
      NULL,
      sizeof (GstRTPBaseDepayload),
      0,
      (GInstanceInitFunc) gst_rtp_base_depayload_init,
    };
    GType _type;

    _type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload",
        &rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT);

    private_offset =
        g_type_add_instance_private (_type,
        sizeof (GstRTPBaseDepayloadPrivate));

    g_once_init_leave ((gsize *) & rtp_base_depayload_type, _type);
  }
  return rtp_base_depayload_type;
}

static inline GstRTPBaseDepayloadPrivate *
gst_rtp_base_depayload_get_instance_private (GstRTPBaseDepayload * self)
{
  return (G_STRUCT_MEMBER_P (self, private_offset));
}

static void
gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;

  gobject_class = G_OBJECT_CLASS (klass);
  gstelement_class = (GstElementClass *) klass;
  parent_class = g_type_class_peek_parent (klass);

  if (private_offset != 0)
    g_type_class_adjust_private_offset (klass, &private_offset);

  gobject_class->finalize = gst_rtp_base_depayload_finalize;
  gobject_class->set_property = gst_rtp_base_depayload_set_property;
  gobject_class->get_property = gst_rtp_base_depayload_get_property;


  /**
   * GstRTPBaseDepayload:stats:
   *
   * Various depayloader statistics retrieved atomically (and are therefore
   * synchroized with each other). This property return a GstStructure named
   * application/x-rtp-depayload-stats containing the following fields relating to
   * the last processed buffer and current state of the stream being depayloaded:
   *
   *   * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream
   *   * `npt-start`: #G_TYPE_UINT64, time of playback start
   *   * `npt-stop`: #G_TYPE_UINT64, time of playback stop
   *   * `play-speed`: #G_TYPE_DOUBLE, the playback speed
   *   * `play-scale`: #G_TYPE_DOUBLE, the playback scale
   *   * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
   *      last DTS
   *   * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
   *      last PTS
   *   * `seqnum`: #G_TYPE_UINT, the last seen seqnum
   *   * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
   **/
  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
      g_param_spec_boxed ("stats", "Statistics", "Various statistics",
          GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  /**
   * GstRTPBaseDepayload:source-info:
   *
   * Add RTP source information found in RTP header as meta to output buffer.
   *
   * Since: 1.16
   **/
  g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
      g_param_spec_boolean ("source-info", "RTP source information",
          "Add RTP source information as buffer meta",
          DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));

  gstelement_class->change_state = gst_rtp_base_depayload_change_state;

  klass->packet_lost = gst_rtp_base_depayload_packet_lost;
  klass->handle_event = gst_rtp_base_depayload_handle_event;

  GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0,
      "Base class for RTP Depayloaders");
}

static void
gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
    GstRTPBaseDepayloadClass * klass)
{
  GstPadTemplate *pad_template;
  GstRTPBaseDepayloadPrivate *priv;

  priv = gst_rtp_base_depayload_get_instance_private (filter);

  filter->priv = priv;

  GST_DEBUG_OBJECT (filter, "init");

  pad_template =
      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
  g_return_if_fail (pad_template != NULL);
  filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
  gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
  gst_pad_set_chain_list_function (filter->sinkpad,
      gst_rtp_base_depayload_chain_list);
  gst_pad_set_event_function (filter->sinkpad,
      gst_rtp_base_depayload_handle_sink_event);
  gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);

  pad_template =
      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
  g_return_if_fail (pad_template != NULL);
  filter->srcpad = gst_pad_new_from_template (pad_template, "src");
  gst_pad_use_fixed_caps (filter->srcpad);
  gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);

  priv->npt_start = 0;
  priv->npt_stop = -1;
  priv->play_speed = 1.0;
  priv->play_scale = 1.0;
  priv->clock_base = -1;
  priv->dts = -1;
  priv->pts = -1;
  priv->duration = -1;
  priv->source_info = DEFAULT_SOURCE_INFO;

  gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
}

static void
gst_rtp_base_depayload_finalize (GObject * object)
{
  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static gboolean
gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
  GstRTPBaseDepayloadClass *bclass;
  GstRTPBaseDepayloadPrivate *priv;
  gboolean res;
  GstStructure *caps_struct;
  const GValue *value;

  priv = filter->priv;

  bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);

  GST_DEBUG_OBJECT (filter, "Set caps %" GST_PTR_FORMAT, caps);

  if (priv->last_caps) {
    if (gst_caps_is_equal (priv->last_caps, caps)) {
      res = TRUE;
      goto caps_not_changed;
    } else {
      gst_caps_unref (priv->last_caps);
      priv->last_caps = NULL;
    }
  }

  caps_struct = gst_caps_get_structure (caps, 0);

  /* get other values for newsegment */
  value = gst_structure_get_value (caps_struct, "npt-start");
  if (value && G_VALUE_HOLDS_UINT64 (value))
    priv->npt_start = g_value_get_uint64 (value);
  else
    priv->npt_start = 0;
  GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);

  value = gst_structure_get_value (caps_struct, "npt-stop");
  if (value && G_VALUE_HOLDS_UINT64 (value))
    priv->npt_stop = g_value_get_uint64 (value);
  else
    priv->npt_stop = -1;

  GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);

  value = gst_structure_get_value (caps_struct, "play-speed");
  if (value && G_VALUE_HOLDS_DOUBLE (value))
    priv->play_speed = g_value_get_double (value);
  else
    priv->play_speed = 1.0;

  value = gst_structure_get_value (caps_struct, "play-scale");
  if (value && G_VALUE_HOLDS_DOUBLE (value))
    priv->play_scale = g_value_get_double (value);
  else
    priv->play_scale = 1.0;

  value = gst_structure_get_value (caps_struct, "clock-base");
  if (value && G_VALUE_HOLDS_UINT (value))
    priv->clock_base = g_value_get_uint (value);
  else
    priv->clock_base = -1;

  if (bclass->set_caps) {
    res = bclass->set_caps (filter, caps);
    if (!res) {
      GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
          caps);
    }
  } else {
    res = TRUE;
  }

  priv->negotiated = res;

  if (priv->negotiated)
    priv->last_caps = gst_caps_ref (caps);

  return res;

caps_not_changed:
  {
    GST_DEBUG_OBJECT (filter, "Caps did not change");
    return res;
  }
}

/* takes ownership of the input buffer */
static GstFlowReturn
gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
    GstRTPBaseDepayloadClass * bclass, GstBuffer * in)
{
  GstBuffer *(*process_rtp_packet_func) (GstRTPBaseDepayload * base,
      GstRTPBuffer * rtp_buffer);
  GstBuffer *(*process_func) (GstRTPBaseDepayload * base, GstBuffer * in);
  GstRTPBaseDepayloadPrivate *priv;
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *out_buf;
  guint32 ssrc;
  guint16 seqnum;
  guint32 rtptime;
  gboolean discont, buf_discont;
  gint gap;
  GstRTPBuffer rtp = { NULL };

  priv = filter->priv;

  process_func = bclass->process;
  process_rtp_packet_func = bclass->process_rtp_packet;

  /* we must have a setcaps first */
  if (G_UNLIKELY (!priv->negotiated))
    goto not_negotiated;

  if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
    goto invalid_buffer;

  buf_discont = GST_BUFFER_IS_DISCONT (in);

  priv->pts = GST_BUFFER_PTS (in);
  priv->dts = GST_BUFFER_DTS (in);
  priv->duration = GST_BUFFER_DURATION (in);

  ssrc = gst_rtp_buffer_get_ssrc (&rtp);
  seqnum = gst_rtp_buffer_get_seq (&rtp);
  rtptime = gst_rtp_buffer_get_timestamp (&rtp);

  priv->last_seqnum = seqnum;
  priv->last_rtptime = rtptime;

  discont = buf_discont;

  GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %"
      GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime,
      GST_TIME_ARGS (priv->pts), GST_TIME_ARGS (priv->dts));

  /* Check seqnum. This is a very simple check that makes sure that the seqnums
   * are strictly increasing, dropping anything that is out of the ordinary. We
   * can only do this when the next_seqnum is known. */
  if (G_LIKELY (priv->next_seqnum != -1)) {
    if (ssrc != priv->last_ssrc) {
      GST_LOG_OBJECT (filter,
          "New ssrc %u (current ssrc %u), sender restarted",
          ssrc, priv->last_ssrc);
      discont = TRUE;
    } else {
      gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);

      /* if we have no gap, all is fine */
      if (G_UNLIKELY (gap != 0)) {
        GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
            priv->next_seqnum, gap);
        if (gap < 0) {
          /* seqnum > next_seqnum, we are missing some packets, this is always a
           * DISCONT. */
          GST_LOG_OBJECT (filter, "%d missing packets", gap);
          discont = TRUE;
        } else {
          /* seqnum < next_seqnum, we have seen this packet before or the sender
           * could be restarted. If the packet is not too old, we throw it away as
           * a duplicate, otherwise we mark discont and continue. 100 misordered
           * packets is a good threshold. See also RFC 4737. */
          if (gap < 100)
            goto dropping;

          GST_LOG_OBJECT (filter,
              "%d > 100, packet too old, sender likely restarted", gap);
          discont = TRUE;
        }
      }
    }
  }
  priv->next_seqnum = (seqnum + 1) & 0xffff;
  priv->last_ssrc = ssrc;

  if (G_UNLIKELY (discont)) {
    priv->discont = TRUE;
    if (!buf_discont) {
      gpointer old_inbuf = in;

      /* we detected a seqnum discont but the buffer was not flagged with a discont,
       * set the discont flag so that the subclass can throw away old data. */
      GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
      in = gst_buffer_make_writable (in);
      GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
      /* depayloaders will check flag on rtpbuffer->buffer, so if the input
       * buffer was not writable already we need to remap to make our
       * newly-flagged buffer current on the rtpbuffer */
      if (in != old_inbuf) {
        gst_rtp_buffer_unmap (&rtp);
        if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
          goto invalid_buffer;
      }
    }
  }

  /* prepare segment event if needed */
  if (filter->need_newsegment) {
    priv->segment_event = create_segment_event (filter, rtptime,
        GST_BUFFER_PTS (in));
    filter->need_newsegment = FALSE;
  }

  priv->input_buffer = in;

  if (process_rtp_packet_func != NULL) {
    out_buf = process_rtp_packet_func (filter, &rtp);
    gst_rtp_buffer_unmap (&rtp);
  } else if (process_func != NULL) {
    gst_rtp_buffer_unmap (&rtp);
    out_buf = process_func (filter, in);
  } else {
    goto no_process;
  }

  /* let's send it out to processing */
  if (out_buf) {
    ret = gst_rtp_base_depayload_push (filter, out_buf);
  }

  gst_buffer_unref (in);
  priv->input_buffer = NULL;

  return ret;

  /* ERRORS */
not_negotiated:
  {
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
        ("No RTP format was negotiated."),
        ("Input buffers need to have RTP caps set on them. This is usually "
            "achieved by setting the 'caps' property of the upstream source "
            "element (often udpsrc or appsrc), or by putting a capsfilter "
            "element before the depayloader and setting the 'caps' property "
            "on that. Also see http://cgit.freedesktop.org/gstreamer/"
            "gst-plugins-good/tree/gst/rtp/README"));
    gst_buffer_unref (in);
    return GST_FLOW_NOT_NEGOTIATED;
  }
invalid_buffer:
  {
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
        ("Received invalid RTP payload, dropping"));
    gst_buffer_unref (in);
    return GST_FLOW_OK;
  }
dropping:
  {
    gst_rtp_buffer_unmap (&rtp);
    GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
    gst_buffer_unref (in);
    return GST_FLOW_OK;
  }
no_process:
  {
    gst_rtp_buffer_unmap (&rtp);
    /* this is not fatal but should be filtered earlier */
    GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
        ("The subclass does not have a process or process_rtp_packet method"));
    gst_buffer_unref (in);
    return GST_FLOW_ERROR;
  }
}

static GstFlowReturn
gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
  GstRTPBaseDepayloadClass *bclass;
  GstRTPBaseDepayload *basedepay;
  GstFlowReturn flow_ret;

  basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);

  bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);

  flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, in);

  return flow_ret;
}

static GstFlowReturn
gst_rtp_base_depayload_chain_list (GstPad * pad, GstObject * parent,
    GstBufferList * list)
{
  GstRTPBaseDepayloadClass *bclass;
  GstRTPBaseDepayload *basedepay;
  GstFlowReturn flow_ret;
  GstBuffer *buffer;
  guint i, len;

  basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);

  bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);

  flow_ret = GST_FLOW_OK;

  /* chain each buffer in list individually */
  len = gst_buffer_list_length (list);

  if (len == 0)
    goto done;

  for (i = 0; i < len; i++) {
    buffer = gst_buffer_list_get (list, i);

    /* handle_buffer takes ownership of input buffer */
    /* FIXME: add a way to steal buffers from list as we will unref it anyway */
    gst_buffer_ref (buffer);

    /* Should we fix up any missing timestamps for list buffers here
     * (e.g. set to first or previous timestamp in list) or just assume
     * the's a jitterbuffer that will have done that for us? */
    flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, buffer);
    if (flow_ret != GST_FLOW_OK)
      break;
  }

done:

  gst_buffer_list_unref (list);

  return flow_ret;
}

static gboolean
gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter,
    GstEvent * event)
{
  gboolean res = TRUE;
  gboolean forward = TRUE;

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_FLUSH_STOP:
      GST_OBJECT_LOCK (filter);
      gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
      GST_OBJECT_UNLOCK (filter);

      filter->need_newsegment = TRUE;
      filter->priv->next_seqnum = -1;
      gst_event_replace (&filter->priv->segment_event, NULL);
      break;
    case GST_EVENT_CAPS:
    {
      GstCaps *caps;

      gst_event_parse_caps (event, &caps);

      res = gst_rtp_base_depayload_setcaps (filter, caps);
      forward = FALSE;
      break;
    }
    case GST_EVENT_SEGMENT:
    {
      GstSegment segment;

      GST_OBJECT_LOCK (filter);
      gst_event_copy_segment (event, &segment);

      if (segment.format != GST_FORMAT_TIME) {
        GST_ERROR_OBJECT (filter, "Segment with non-TIME format not supported");
        res = FALSE;
      }
      filter->priv->segment_seqnum = gst_event_get_seqnum (event);
      filter->segment = segment;
      GST_OBJECT_UNLOCK (filter);

      /* don't pass the event downstream, we generate our own segment including
       * the NTP time and other things we receive in caps */
      forward = FALSE;
      break;
    }
    case GST_EVENT_CUSTOM_DOWNSTREAM:
    {
      GstRTPBaseDepayloadClass *bclass;

      bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);

      if (gst_event_has_name (event, "GstRTPPacketLost")) {
        /* we get this event from the jitterbuffer when it considers a packet as
         * being lost. We send it to our packet_lost vmethod. The default
         * implementation will make time progress by pushing out a GAP event.
         * Subclasses can override and do one of the following:
         *  - Adjust timestamp/duration to something more accurate before
         *    calling the parent (default) packet_lost method.
         *  - do some more advanced error concealing on the already received
         *    (fragmented) packets.
         *  - ignore the packet lost.
         */
        if (bclass->packet_lost)
          res = bclass->packet_lost (filter, event);
        forward = FALSE;
      }
      break;
    }
    default:
      break;
  }

  if (forward)
    res = gst_pad_push_event (filter->srcpad, event);
  else
    gst_event_unref (event);

  return res;
}

static gboolean
gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  gboolean res = FALSE;
  GstRTPBaseDepayload *filter;
  GstRTPBaseDepayloadClass *bclass;

  filter = GST_RTP_BASE_DEPAYLOAD (parent);
  bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
  if (bclass->handle_event)
    res = bclass->handle_event (filter, event);
  else
    gst_event_unref (event);

  return res;
}

static GstEvent *
create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
    GstClockTime position)
{
  GstEvent *event;
  GstClockTime start, stop, running_time;
  GstRTPBaseDepayloadPrivate *priv;
  GstSegment segment;

  priv = filter->priv;

  /* We don't need the object lock around - the segment
   * can't change here while we're holding the STREAM_LOCK
   */

  /* determining the start of the segment */
  start = filter->segment.start;
  if (priv->clock_base != -1 && position != -1) {
    GstClockTime exttime, gap;

    exttime = priv->clock_base;
    gst_rtp_buffer_ext_timestamp (&exttime, rtptime);
    gap = gst_util_uint64_scale_int (exttime - priv->clock_base,
        filter->clock_rate, GST_SECOND);

    /* account for lost packets */
    if (position > gap) {
      GST_DEBUG_OBJECT (filter,
          "Found gap of %" GST_TIME_FORMAT ", adjusting start: %"
          GST_TIME_FORMAT " = %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
          GST_TIME_ARGS (gap), GST_TIME_ARGS (position - gap),
          GST_TIME_ARGS (position), GST_TIME_ARGS (gap));
      start = position - gap;
    }
  }

  /* determining the stop of the segment */
  stop = filter->segment.stop;
  if (priv->npt_stop != -1)
    stop = start + (priv->npt_stop - priv->npt_start);

  if (position == -1)
    position = start;

  running_time = gst_segment_to_running_time (&filter->segment,
      GST_FORMAT_TIME, start);

  gst_segment_init (&segment, GST_FORMAT_TIME);
  segment.rate = priv->play_speed;
  segment.applied_rate = priv->play_scale;
  segment.start = start;
  segment.stop = stop;
  segment.time = priv->npt_start;
  segment.position = position;
  segment.base = running_time;

  GST_DEBUG_OBJECT (filter, "Creating segment event %" GST_SEGMENT_FORMAT,
      &segment);
  event = gst_event_new_segment (&segment);
  if (filter->priv->segment_seqnum != GST_SEQNUM_INVALID)
    gst_event_set_seqnum (event, filter->priv->segment_seqnum);

  return event;
}

static void
add_rtp_source_meta (GstBuffer * outbuf, GstBuffer * rtpbuf)
{
  GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
  GstRTPSourceMeta *meta;
  guint32 ssrc;

  if (!gst_rtp_buffer_map (rtpbuf, GST_MAP_READ, &rtp))
    return;

  ssrc = gst_rtp_buffer_get_ssrc (&rtp);
  meta = gst_buffer_add_rtp_source_meta (outbuf, &ssrc, NULL, 0);
  if (meta != NULL) {
    gint i;
    gint csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
    for (i = 0; i < csrc_count; i++) {
      guint32 csrc = gst_rtp_buffer_get_csrc (&rtp, i);
      gst_rtp_source_meta_append_csrc (meta, &csrc, 1);
    }
  }

  gst_rtp_buffer_unmap (&rtp);
}

static gboolean
set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
{
  GstRTPBaseDepayloadPrivate *priv = depayload->priv;
  GstClockTime pts, dts, duration;

  *buffer = gst_buffer_make_writable (*buffer);

  pts = GST_BUFFER_PTS (*buffer);
  dts = GST_BUFFER_DTS (*buffer);
  duration = GST_BUFFER_DURATION (*buffer);

  /* apply last incomming timestamp and duration to outgoing buffer if
   * not otherwise set. */
  if (!GST_CLOCK_TIME_IS_VALID (pts))
    GST_BUFFER_PTS (*buffer) = priv->pts;
  if (!GST_CLOCK_TIME_IS_VALID (dts))
    GST_BUFFER_DTS (*buffer) = priv->dts;
  if (!GST_CLOCK_TIME_IS_VALID (duration))
    GST_BUFFER_DURATION (*buffer) = priv->duration;

  if (G_UNLIKELY (depayload->priv->discont)) {
    GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
    GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
    depayload->priv->discont = FALSE;
  }

  /* make sure we only set the timestamp on the first packet */
  priv->pts = GST_CLOCK_TIME_NONE;
  priv->dts = GST_CLOCK_TIME_NONE;
  priv->duration = GST_CLOCK_TIME_NONE;

  if (priv->source_info && priv->input_buffer)
    add_rtp_source_meta (*buffer, priv->input_buffer);

  return TRUE;
}

static GstFlowReturn
gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter,
    gboolean is_list, gpointer obj)
{
  if (is_list) {
    GstBufferList **blist = obj;
    gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, filter);
  } else {
    GstBuffer **buf = obj;
    set_headers (buf, 0, filter);
  }

  /* if this is the first buffer send a NEWSEGMENT */
  if (G_UNLIKELY (filter->priv->segment_event)) {
    gst_pad_push_event (filter->srcpad, filter->priv->segment_event);
    filter->priv->segment_event = NULL;
    GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
  }

  return GST_FLOW_OK;
}

/**
 * gst_rtp_base_depayload_push:
 * @filter: a #GstRTPBaseDepayload
 * @out_buf: a #GstBuffer
 *
 * Push @out_buf to the peer of @filter. This function takes ownership of
 * @out_buf.
 *
 * This function will by default apply the last incomming timestamp on
 * the outgoing buffer when it didn't have a timestamp already.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
{
  GstFlowReturn res;

  res = gst_rtp_base_depayload_prepare_push (filter, FALSE, &out_buf);

  if (G_LIKELY (res == GST_FLOW_OK))
    res = gst_pad_push (filter->srcpad, out_buf);
  else
    gst_buffer_unref (out_buf);

  return res;
}

/**
 * gst_rtp_base_depayload_push_list:
 * @filter: a #GstRTPBaseDepayload
 * @out_list: a #GstBufferList
 *
 * Push @out_list to the peer of @filter. This function takes ownership of
 * @out_list.
 *
 * Returns: a #GstFlowReturn.
 */
GstFlowReturn
gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
    GstBufferList * out_list)
{
  GstFlowReturn res;

  res = gst_rtp_base_depayload_prepare_push (filter, TRUE, &out_list);

  if (G_LIKELY (res == GST_FLOW_OK))
    res = gst_pad_push_list (filter->srcpad, out_list);
  else
    gst_buffer_list_unref (out_list);

  return res;
}

/* convert the PacketLost event from a jitterbuffer to a GAP event.
 * subclasses can override this.  */
static gboolean
gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter,
    GstEvent * event)
{
  GstClockTime timestamp, duration;
  GstEvent *sevent;
  const GstStructure *s;
  gboolean might_have_been_fec;
  gboolean res = TRUE;

  s = gst_event_get_structure (event);

  /* first start by parsing the timestamp and duration */
  timestamp = -1;
  duration = -1;

  if (!gst_structure_get_clock_time (s, "timestamp", &timestamp) ||
      !gst_structure_get_clock_time (s, "duration", &duration)) {
    GST_ERROR_OBJECT (filter,
        "Packet loss event without timestamp or duration");
    return FALSE;
  }

  sevent = gst_pad_get_sticky_event (filter->srcpad, GST_EVENT_SEGMENT, 0);
  if (G_UNLIKELY (!sevent)) {
    /* Typically happens if lost event arrives before first buffer */
    GST_DEBUG_OBJECT (filter,
        "Ignore packet loss because segment event missing");
    return FALSE;
  }
  gst_event_unref (sevent);

  if (!gst_structure_get_boolean (s, "might-have-been-fec",
          &might_have_been_fec) || !might_have_been_fec) {
    /* send GAP event */
    sevent = gst_event_new_gap (timestamp, duration);
    res = gst_pad_push_event (filter->srcpad, sevent);
  }

  return res;
}

static GstStateChangeReturn
gst_rtp_base_depayload_change_state (GstElement * element,
    GstStateChange transition)
{
  GstRTPBaseDepayload *filter;
  GstRTPBaseDepayloadPrivate *priv;
  GstStateChangeReturn ret;

  filter = GST_RTP_BASE_DEPAYLOAD (element);
  priv = filter->priv;

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      filter->need_newsegment = TRUE;
      priv->npt_start = 0;
      priv->npt_stop = -1;
      priv->play_speed = 1.0;
      priv->play_scale = 1.0;
      priv->clock_base = -1;
      priv->next_seqnum = -1;
      priv->negotiated = FALSE;
      priv->discont = FALSE;
      priv->segment_seqnum = GST_SEQNUM_INVALID;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      break;
    default:
      break;
  }

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

  switch (transition) {
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      gst_caps_replace (&priv->last_caps, NULL);
      gst_event_replace (&priv->segment_event, NULL);
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      break;
    default:
      break;
  }
  return ret;
}

static GstStructure *
gst_rtp_base_depayload_create_stats (GstRTPBaseDepayload * depayload)
{
  GstRTPBaseDepayloadPrivate *priv;
  GstStructure *s;
  GstClockTime pts = GST_CLOCK_TIME_NONE, dts = GST_CLOCK_TIME_NONE;

  priv = depayload->priv;

  GST_OBJECT_LOCK (depayload);
  if (depayload->segment.format != GST_FORMAT_UNDEFINED) {
    pts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
        priv->pts);
    dts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
        priv->dts);
  }
  GST_OBJECT_UNLOCK (depayload);

  s = gst_structure_new ("application/x-rtp-depayload-stats",
      "clock_rate", G_TYPE_UINT, depayload->clock_rate,
      "npt-start", G_TYPE_UINT64, priv->npt_start,
      "npt-stop", G_TYPE_UINT64, priv->npt_stop,
      "play-speed", G_TYPE_DOUBLE, priv->play_speed,
      "play-scale", G_TYPE_DOUBLE, priv->play_scale,
      "running-time-dts", G_TYPE_UINT64, dts,
      "running-time-pts", G_TYPE_UINT64, pts,
      "seqnum", G_TYPE_UINT, (guint) priv->last_seqnum,
      "timestamp", G_TYPE_UINT, (guint) priv->last_rtptime, NULL);

  return s;
}


static void
gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstRTPBaseDepayload *depayload;

  depayload = GST_RTP_BASE_DEPAYLOAD (object);

  switch (prop_id) {
    case PROP_SOURCE_INFO:
      gst_rtp_base_depayload_set_source_info_enabled (depayload,
          g_value_get_boolean (value));
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstRTPBaseDepayload *depayload;

  depayload = GST_RTP_BASE_DEPAYLOAD (object);

  switch (prop_id) {
    case PROP_STATS:
      g_value_take_boxed (value,
          gst_rtp_base_depayload_create_stats (depayload));
      break;
    case PROP_SOURCE_INFO:
      g_value_set_boolean (value,
          gst_rtp_base_depayload_is_source_info_enabled (depayload));
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

/**
 * gst_rtp_base_depayload_set_source_info_enabled:
 * @depayload: a #GstRTPBaseDepayload
 * @enable: whether to add meta about RTP sources to buffer
 *
 * Enable or disable adding #GstRTPSourceMeta to depayloaded buffers.
 *
 * Since: 1.16
 **/
void
gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
    gboolean enable)
{
  depayload->priv->source_info = enable;
}

/**
 * gst_rtp_base_depayload_is_source_info_enabled:
 * @depayload: a #GstRTPBaseDepayload
 *
 * Queries whether #GstRTPSourceMeta will be added to depayloaded buffers.
 *
 * Returns: %TRUE if source-info is enabled.
 *
 * Since: 1.16
 **/
gboolean
gst_rtp_base_depayload_is_source_info_enabled (GstRTPBaseDepayload * depayload)
{
  return depayload->priv->source_info;
}
