--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/rtp/gstbasertpdepayload.c	Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,531 @@
+/* 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:gstbasertpdepayload
+ * @short_description: Base class for RTP depayloader
+ *
+ * <refsect2>
+ * <para>
+ * Provides a base class for RTP depayloaders
+ * </para>
+ * </refsect2>
+ */
+
+#include "gstbasertpdepayload.h"
+
+#ifdef GST_DISABLE_DEPRECATED
+#define QUEUE_LOCK_INIT(base)   (g_static_rec_mutex_init(&base->queuelock))
+#define QUEUE_LOCK_FREE(base)   (g_static_rec_mutex_free(&base->queuelock))
+#define QUEUE_LOCK(base)        (g_static_rec_mutex_lock(&base->queuelock))
+#define QUEUE_UNLOCK(base)      (g_static_rec_mutex_unlock(&base->queuelock))
+#else
+/* otherwise it's already been defined in the header (FIXME 0.11)*/
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
+#define GST_CAT_DEFAULT (basertpdepayload_debug)
+
+#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
+
+struct _GstBaseRTPDepayloadPrivate
+{
+  GstClockTime npt_start;
+  GstClockTime npt_stop;
+  gdouble play_speed;
+  gdouble play_scale;
+
+  gboolean discont;
+  GstClockTime timestamp;
+  GstClockTime duration;
+};
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+#define DEFAULT_QUEUE_DELAY	0
+
+enum
+{
+  PROP_0,
+  PROP_QUEUE_DELAY
+};
+
+static void gst_base_rtp_depayload_finalize (GObject * object);
+static void gst_base_rtp_depayload_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_rtp_depayload_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
+static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
+    GstBuffer * in);
+static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
+    GstEvent * event);
+
+static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
+    element, GstStateChange transition);
+
+static void gst_base_rtp_depayload_set_gst_timestamp
+    (GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
+
+GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
+    GST_TYPE_ELEMENT);
+
+static void
+gst_base_rtp_depayload_base_init (gpointer klass)
+{
+  /*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
+}
+
+static void
+gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * 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);
+
+  g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
+
+  gobject_class->finalize = gst_base_rtp_depayload_finalize;
+  gobject_class->set_property = gst_base_rtp_depayload_set_property;
+  gobject_class->get_property = gst_base_rtp_depayload_get_property;
+
+  /**
+   * GstBaseRTPDepayload::queue-delay
+   *
+   * Control the amount of packets to buffer.
+   *
+   * Deprecated: Use a jitterbuffer or RTP session manager to delay packet
+   * playback. This property has no effect anymore since 0.10.15.
+   */
+#ifndef GST_REMOVE_DEPRECATED
+  g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
+      g_param_spec_uint ("queue-delay", "Queue Delay",
+          "Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
+          DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE));
+#endif
+
+  gstelement_class->change_state = gst_base_rtp_depayload_change_state;
+
+  klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
+
+  GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
+      "Base class for RTP Depayloaders");
+}
+
+static void
+gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
+    GstBaseRTPDepayloadClass * klass)
+{
+  GstPadTemplate *pad_template;
+  GstBaseRTPDepayloadPrivate *priv;
+
+  priv = GST_BASE_RTP_DEPAYLOAD_GET_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_setcaps_function (filter->sinkpad,
+      gst_base_rtp_depayload_setcaps);
+  gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
+  gst_pad_set_event_function (filter->sinkpad,
+      gst_base_rtp_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);
+
+  filter->queue = g_queue_new ();
+  filter->queue_delay = DEFAULT_QUEUE_DELAY;
+
+  gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
+}
+
+static void
+gst_base_rtp_depayload_finalize (GObject * object)
+{
+  GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+  g_queue_free (filter->queue);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstBaseRTPDepayload *filter;
+  GstBaseRTPDepayloadClass *bclass;
+  GstBaseRTPDepayloadPrivate *priv;
+  gboolean res;
+  GstStructure *caps_struct;
+  const GValue *value;
+
+  filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
+  priv = filter->priv;
+
+  bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+  GST_DEBUG_OBJECT (filter, "Set caps");
+
+  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;
+
+  if (bclass->set_caps)
+    res = bclass->set_caps (filter, caps);
+  else
+    res = TRUE;
+
+  gst_object_unref (filter);
+
+  return res;
+}
+
+static GstFlowReturn
+gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
+{
+  GstBaseRTPDepayload *filter;
+  GstBaseRTPDepayloadPrivate *priv;
+  GstBaseRTPDepayloadClass *bclass;
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstBuffer *out_buf;
+  GstClockTime timestamp;
+
+  filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
+
+  priv = filter->priv;
+  priv->discont = GST_BUFFER_IS_DISCONT (in);
+
+  /* convert to running_time and save the timestamp, this is the timestamp
+   * we put on outgoing buffers. */
+  timestamp = GST_BUFFER_TIMESTAMP (in);
+  timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
+      timestamp);
+  priv->timestamp = timestamp;
+  priv->duration = GST_BUFFER_DURATION (in);
+
+  bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+  /* let's send it out to processing */
+  out_buf = bclass->process (filter, in);
+  if (out_buf) {
+    guint32 rtptime;
+
+    rtptime = gst_rtp_buffer_get_timestamp (in);
+
+    /* we pass rtptime as backward compatibility, in reality, the incomming
+     * buffer timestamp is always applied to the outgoing packet. */
+    ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
+  }
+  gst_buffer_unref (in);
+
+  return ret;
+}
+
+static gboolean
+gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstBaseRTPDepayload *filter;
+  gboolean res = TRUE;
+
+  filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_FLUSH_STOP:
+      res = gst_pad_push_event (filter->srcpad, event);
+
+      gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
+      filter->need_newsegment = TRUE;
+      break;
+    case GST_EVENT_NEWSEGMENT:
+    {
+      gboolean update;
+      gdouble rate;
+      GstFormat fmt;
+      gint64 start, stop, position;
+
+      gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
+          &position);
+
+      gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
+          start, stop, position);
+
+      /* don't pass the event downstream, we generate our own segment including
+       * the NTP time and other things we receive in caps */
+      gst_event_unref (event);
+      break;
+    }
+    default:
+      /* pass other events forward */
+      res = gst_pad_push_event (filter->srcpad, event);
+      break;
+  }
+  return res;
+}
+
+static GstFlowReturn
+gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
+    gboolean do_ts, guint32 rtptime, GstBuffer * out_buf)
+{
+  GstFlowReturn ret;
+  GstCaps *srccaps;
+  GstBaseRTPDepayloadClass *bclass;
+  GstBaseRTPDepayloadPrivate *priv;
+
+  priv = filter->priv;
+
+  /* set the caps if any */
+  srccaps = GST_PAD_CAPS (filter->srcpad);
+  if (srccaps)
+    gst_buffer_set_caps (out_buf, srccaps);
+
+  bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+  /* set the timestamp if we must and can */
+  if (bclass->set_gst_timestamp && do_ts)
+    bclass->set_gst_timestamp (filter, rtptime, out_buf);
+
+  if (priv->discont) {
+    GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
+    priv->discont = FALSE;
+  }
+
+  /* push it */
+  GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
+      GST_BUFFER_SIZE (out_buf),
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)));
+  ret = gst_pad_push (filter->srcpad, out_buf);
+  GST_LOG_OBJECT (filter, "Pushed buffer: %s", gst_flow_get_name (ret));
+
+  return ret;
+}
+
+/**
+ * gst_base_rtp_depayload_push_ts:
+ * @filter: a #GstBaseRTPDepayload
+ * @timestamp: an RTP timestamp to apply
+ * @out_buf: a #GstBuffer
+ *
+ * Push @out_buf to the peer of @filter. This function takes ownership of
+ * @out_buf.
+ *
+ * Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp
+ * on the outgoing buffer, using the configured clock_rate to convert the
+ * timestamp to a valid GStreamer clock time.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstFlowReturn
+gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
+    GstBuffer * out_buf)
+{
+  return gst_base_rtp_depayload_push_full (filter, TRUE, timestamp, out_buf);
+}
+
+/**
+ * gst_base_rtp_depayload_push:
+ * @filter: a #GstBaseRTPDepayload
+ * @out_buf: a #GstBuffer
+ *
+ * Push @out_buf to the peer of @filter. This function takes ownership of
+ * @out_buf.
+ *
+ * Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
+ * any timestamp on the outgoing buffer.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstFlowReturn
+gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
+{
+  return gst_base_rtp_depayload_push_full (filter, FALSE, 0, out_buf);
+}
+
+static void
+gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
+    guint32 rtptime, GstBuffer * buf)
+{
+  GstBaseRTPDepayloadPrivate *priv;
+  GstClockTime timestamp, duration;
+
+  priv = filter->priv;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  duration = GST_BUFFER_DURATION (buf);
+
+  /* apply last incomming timestamp and duration to outgoing buffer if
+   * not otherwise set. */
+  if (!GST_CLOCK_TIME_IS_VALID (timestamp))
+    GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
+  if (!GST_CLOCK_TIME_IS_VALID (duration))
+    GST_BUFFER_DURATION (buf) = priv->duration;
+
+  /* if this is the first buffer send a NEWSEGMENT */
+  if (filter->need_newsegment) {
+    GstEvent *event;
+    GstClockTime stop, position;
+
+    if (priv->npt_stop != -1)
+      stop = priv->npt_stop - priv->npt_start;
+    else
+      stop = -1;
+
+    position = priv->npt_start;
+
+    event =
+        gst_event_new_new_segment_full (FALSE, priv->play_speed,
+        priv->play_scale, GST_FORMAT_TIME, 0, stop, position);
+
+    gst_pad_push_event (filter->srcpad, event);
+
+    filter->need_newsegment = FALSE;
+    GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
+  }
+}
+
+static GstStateChangeReturn
+gst_base_rtp_depayload_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstBaseRTPDepayload *filter;
+  GstBaseRTPDepayloadPrivate *priv;
+  GstStateChangeReturn ret;
+
+  filter = GST_BASE_RTP_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;
+      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:
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+  return ret;
+}
+
+static void
+gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstBaseRTPDepayload *filter;
+
+  filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+  switch (prop_id) {
+    case PROP_QUEUE_DELAY:
+      filter->queue_delay = g_value_get_uint (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstBaseRTPDepayload *filter;
+
+  filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+  switch (prop_id) {
+    case PROP_QUEUE_DELAY:
+      g_value_set_uint (value, filter->queue_delay);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}