rtpbin: add support for buffer-list

Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.

fixes #585839

gst/rtpmanager/gstrtpsession.c

@@ -259,7 +259,7 @@ struct _GstRtpSessionPrivate
 static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
     RTPSource * src, GstBuffer * buffer, gpointer user_data);
 static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
-    RTPSource * src, GstBuffer * buffer, gpointer user_data);
+    RTPSource * src, gpointer data, gpointer user_data);
 static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
     RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
 static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
@@ -1032,8 +1032,8 @@ gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
   g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
 }
 
-/* called when the session manager has an RTP packet ready for further
+/* called when the session manager has an RTP packet or a list of packets
- * processing */
+ * ready for further processing */
 static GstFlowReturn
 gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
     GstBuffer * buffer, gpointer user_data)
@@ -1060,7 +1060,7 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
  * sending */
 static GstFlowReturn
 gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
-    GstBuffer * buffer, gpointer user_data)
+    gpointer data, gpointer user_data)
 {
   GstFlowReturn result;
   GstRtpSession *rtpsession;
@@ -1069,12 +1069,17 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
   rtpsession = GST_RTP_SESSION (user_data);
   priv = rtpsession->priv;
 
-  GST_LOG_OBJECT (rtpsession, "sending RTP packet");
-
   if (rtpsession->send_rtp_src) {
-    result = gst_pad_push (rtpsession->send_rtp_src, buffer);
+    if (GST_IS_BUFFER (data)) {
+      GST_LOG_OBJECT (rtpsession, "sending RTP packet");
+      result = gst_pad_push (rtpsession->send_rtp_src, GST_BUFFER_CAST (data));
+    } else {
+      GST_LOG_OBJECT (rtpsession, "sending RTP list");
+      result = gst_pad_push_list (rtpsession->send_rtp_src,
+          GST_BUFFER_LIST_CAST (data));
+    }
   } else {
-    gst_buffer_unref (buffer);
+    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
     result = GST_FLOW_OK;
   }
   return result;
@@ -1642,11 +1647,12 @@ gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps)
   return TRUE;
 }
 
-/* Recieve an RTP packet to be send to the receivers, send to RTP session
+/* Recieve an RTP packet or a list of packets to be send to the receivers,
- * manager and forward to send_rtp_src.
+ * send to RTP session manager and forward to send_rtp_src.
  */
 static GstFlowReturn
-gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
+gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data,
+    gboolean is_list)
 {
   GstRtpSession *rtpsession;
   GstRtpSessionPrivate *priv;
@@ -1658,10 +1664,22 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
   rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
   priv = rtpsession->priv;
 
-  GST_LOG_OBJECT (rtpsession, "received RTP packet");
+  GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
 
   /* get NTP time when this packet was captured, this depends on the timestamp. */
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+  if (is_list) {
+    GstBuffer *buffer = NULL;
+
+    /* All groups in an list have the same timestamp.
+     * So, just take it from the first group. */
+    buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0);
+    if (buffer)
+      timestamp = GST_BUFFER_TIMESTAMP (buffer);
+    else
+      timestamp = -1;
+  } else {
+    timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
+  }
   if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
     /* convert to running time using the segment start value. */
     ntpnstime =
@@ -1676,7 +1694,9 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
   }
 
   current_time = gst_clock_get_time (priv->sysclock);
-  ret = rtp_session_send_rtp (priv->session, buffer, current_time, ntpnstime);
+  ret =
+      rtp_session_send_rtp (priv->session, data, is_list, current_time,
+      ntpnstime);
   if (ret != GST_FLOW_OK)
     goto push_error;
 
@@ -1694,6 +1714,18 @@ push_error:
   }
 }
 
+static GstFlowReturn
+gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
+{
+  return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE);
+}
+
+static GstFlowReturn
+gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list)
+{
+  return gst_rtp_session_chain_send_rtp_common (pad, list, TRUE);
+}
+
 /* Create sinkpad to receive RTP packets from senders. This will also create a
  * srcpad for the RTP packets.
  */
@@ -1817,6 +1849,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession)
       "send_rtp_sink");
   gst_pad_set_chain_function (rtpsession->send_rtp_sink,
       gst_rtp_session_chain_send_rtp);
+  gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
+      gst_rtp_session_chain_send_rtp_list);
   gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
       gst_rtp_session_getcaps_send_rtp);
   gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,

gst/rtpmanager/rtpsession.c

@@ -958,7 +958,7 @@ rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
 }
 
 static GstFlowReturn
-source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
+source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
 {
   GstFlowReturn result = GST_FLOW_OK;
 
@@ -969,21 +969,21 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
 
     if (session->callbacks.send_rtp)
       result =
-          session->callbacks.send_rtp (session, source, buffer,
+          session->callbacks.send_rtp (session, source, data,
           session->send_rtp_user_data);
-    else
+    else {
-      gst_buffer_unref (buffer);
+      gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-
+    }
   } else {
     GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
     RTP_SESSION_UNLOCK (session);
 
     if (session->callbacks.process_rtp)
       result =
-          session->callbacks.process_rtp (session, source, buffer,
+          session->callbacks.process_rtp (session, source,
-          session->process_rtp_user_data);
+          GST_BUFFER_CAST (data), session->process_rtp_user_data);
     else
-      gst_buffer_unref (buffer);
+      gst_buffer_unref (GST_BUFFER_CAST (data));
   }
   RTP_SESSION_LOCK (session);
 
@@ -1962,7 +1962,7 @@ ignore:
 /**
  * rtp_session_send_rtp:
  * @sess: an #RTPSession
- * @buffer: an RTP buffer
+ * @data: pointer to either an RTP buffer or a list of RTP buffers
  * @current_time: the current system time
  * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
  * This is the buffer timestamp converted to NTP time.
@@ -1973,20 +1973,27 @@ ignore:
  * Returns: a #GstFlowReturn.
  */
 GstFlowReturn
-rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
+rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
     GstClockTime current_time, guint64 ntpnstime)
 {
   GstFlowReturn result;
   RTPSource *source;
   gboolean prevsender;
+  gboolean valid_packet;
 
   g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+  g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
 
-  if (!gst_rtp_buffer_validate (buffer))
+  if (is_list) {
+    valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
+  } else {
+    valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
+  }
+
+  if (!valid_packet)
     goto invalid_packet;
 
-  GST_LOG ("received RTP packet for sending");
+  GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
 
   RTP_SESSION_LOCK (sess);
   source = sess->source;
@@ -1997,7 +2004,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
   prevsender = RTP_SOURCE_IS_SENDER (source);
 
   /* we use our own source to send */
-  result = rtp_source_send_rtp (source, buffer, ntpnstime);
+  result = rtp_source_send_rtp (source, data, is_list, ntpnstime);
 
   if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
     sess->stats.sender_sources++;
@@ -2008,7 +2015,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
   /* ERRORS */
 invalid_packet:
   {
-    gst_buffer_unref (buffer);
+    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
     GST_DEBUG ("invalid RTP packet received");
     return GST_FLOW_OK;
   }

gst/rtpmanager/rtpsession.h

@@ -65,7 +65,7 @@ typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src,
  *
  * Returns: a #GstFlowReturn.
  */
-typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
+typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gpointer data, gpointer user_data);
 
 /**
  * RTPSessionSendRTCP:
@@ -288,7 +288,7 @@ GstFlowReturn   rtp_session_process_rtcp           (RTPSession *sess, GstBuffer
                                                     GstClockTime current_time);
 
 /* processing packets for sending */
-GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, GstBuffer *buffer,
+GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, gpointer data, gboolean is_list,
                                                     GstClockTime current_time, guint64 ntpnstime);
 
 /* stopping the session */

gst/rtpmanager/rtpsource.c

@@ -26,7 +26,7 @@
 GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
 #define GST_CAT_DEFAULT rtp_source_debug
 
-#define RTP_MAX_PROBATION_LEN	32
+#define RTP_MAX_PROBATION_LEN  32
 
 /* signals and args */
 enum
@@ -1091,41 +1091,73 @@ rtp_source_process_bye (RTPSource * src, const gchar * reason)
   src->received_bye = TRUE;
 }
 
+static GstBufferListItem
+set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
+{
+  *buffer = gst_buffer_make_writable (*buffer);
+  gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
+  return GST_BUFFER_LIST_SKIP_GROUP;
+}
+
 /**
  * rtp_source_send_rtp:
  * @src: an #RTPSource
- * @buffer: an RTP buffer
+ * @data: an RTP buffer or a list of RTP buffers
+ * @is_list: if @data is a buffer or list
  * @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This
  * is the buffer timestamp converted to NTP time.
  *
- * Send an RTP @buffer originating from @src. This will make @src a sender.
+ * Send @data (an RTP buffer or list of buffers) originating from @src.
- * This function takes ownership of @buffer and modifies the SSRC in the RTP
+ * This will make @src a sender. This function takes ownership of @data and
- * packet to that of @src when needed.
+ * modifies the SSRC in the RTP packet to that of @src when needed.
  *
  * Returns: a #GstFlowReturn.
  */
 GstFlowReturn
-rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
+rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
+    guint64 ntpnstime)
 {
-  GstFlowReturn result = GST_FLOW_OK;
+  GstFlowReturn result;
   guint len;
   guint32 rtptime;
   guint64 ext_rtptime;
   guint64 ntp_diff, rtp_diff;
   guint64 elapsed;
+  GstBufferList *list = NULL;
+  GstBuffer *buffer = NULL;
+  guint packets;
+  guint32 ssrc;
 
   g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+  g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
 
-  len = gst_rtp_buffer_get_payload_len (buffer);
+  if (is_list) {
+    list = GST_BUFFER_LIST_CAST (data);
 
+    /* We can grab the caps from the first group, since all
+     * groups of a buffer list have same caps. */
+    buffer = gst_buffer_list_get (list, 0, 0);
+    if (!buffer)
+      goto no_buffer;
+  } else {
+    buffer = GST_BUFFER_CAST (data);
+  }
   rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
 
   /* we are a sender now */
   src->is_sender = TRUE;
 
+  if (is_list) {
+    /* Each group makes up a network packet. */
+    packets = gst_buffer_list_n_groups (list);
+    len = gst_rtp_buffer_list_get_payload_len (list);
+  } else {
+    packets = 1;
+    len = gst_rtp_buffer_get_payload_len (buffer);
+  }
+
   /* update stats for the SR */
-  src->stats.packets_sent++;
+  src->stats.packets_sent += packets;
   src->stats.octets_sent += len;
   src->bytes_sent += len;
 
@@ -1156,7 +1188,11 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
     src->bitrate = 0;
   }
 
-  rtptime = gst_rtp_buffer_get_timestamp (buffer);
+  if (is_list) {
+    rtptime = gst_rtp_buffer_list_get_timestamp (list);
+  } else {
+    rtptime = gst_rtp_buffer_get_timestamp (buffer);
+  }
   ext_rtptime = src->last_rtptime;
   ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
 
@@ -1180,31 +1216,53 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
   src->last_ntpnstime = ntpnstime;
 
   /* push packet */
-  if (src->callbacks.push_rtp) {
+  if (!src->callbacks.push_rtp)
-    guint32 ssrc;
+    goto no_callback;
 
-    ssrc = gst_rtp_buffer_get_ssrc (buffer);
+  if (is_list) {
-    if (ssrc != src->ssrc) {
+    ssrc = gst_rtp_buffer_list_get_ssrc (list);
-      /* the SSRC of the packet is not correct, make a writable buffer and
-       * update the SSRC. This could involve a complete copy of the packet when
-       * it is not writable. Usually the payloader will use caps negotiation to
-       * get the correct SSRC from the session manager before pushing anything. */
-      buffer = gst_buffer_make_writable (buffer);
-
-      /* FIXME, we don't want to warn yet because we can't inform any payloader
-       * of the changes SSRC yet because we don't implement pad-alloc. */
-      GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
-          src->ssrc);
-      gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
-    }
-    GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
-    result = src->callbacks.push_rtp (src, buffer, src->user_data);
   } else {
-    GST_WARNING ("no callback installed, dropping packet");
+    ssrc = gst_rtp_buffer_get_ssrc (buffer);
-    gst_buffer_unref (buffer);
   }
 
+  if (ssrc != src->ssrc) {
+    /* the SSRC of the packet is not correct, make a writable buffer and
+     * update the SSRC. This could involve a complete copy of the packet when
+     * it is not writable. Usually the payloader will use caps negotiation to
+     * get the correct SSRC from the session manager before pushing anything. */
+
+    /* FIXME, we don't want to warn yet because we can't inform any payloader
+     * of the changes SSRC yet because we don't implement pad-alloc. */
+    GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
+        src->ssrc);
+
+    if (is_list) {
+      list = gst_buffer_list_make_writable (list);
+      gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
+    } else {
+      set_ssrc (&buffer, 0, 0, src);
+    }
+  }
+  GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
+      src->stats.packets_sent);
+
+  result = src->callbacks.push_rtp (src, data, src->user_data);
+
   return result;
+
+  /* ERRORS */
+no_buffer:
+  {
+    GST_WARNING ("no buffers in buffer list");
+    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+    return GST_FLOW_OK;
+  }
+no_callback:
+  {
+    GST_WARNING ("no callback installed, dropping packet");
+    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+    return GST_FLOW_OK;
+  }
 }
 
 /**

gst/rtpmanager/rtpsource.h

@@ -194,7 +194,7 @@ void            rtp_source_set_rtcp_from       (RTPSource *src, GstNetAddress *a
 /* handling RTP */
 GstFlowReturn   rtp_source_process_rtp         (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
 
-GstFlowReturn   rtp_source_send_rtp            (RTPSource *src, GstBuffer *buffer, guint64 ntpnstime);
+GstFlowReturn   rtp_source_send_rtp            (RTPSource *src, gpointer data, gboolean is_list, guint64 ntpnstime);
 
 /* RTCP messages */
 void            rtp_source_process_bye         (RTPSource *src, const gchar *reason);

tests/check/elements/rtpbin_buffer_list.c

@@ -0,0 +1,331 @@
+/* GStreamer
+ *
+ * Unit test for gstrtpbin sending rtp packets using GstBufferList.
+ * Copyright (C) 2009 Branko Subasic <branko dot subasic at axis dot 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.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+
+
+/* This test makes sure that RTP packets sent as buffer lists are sent through
+ * the rtpbin as they are supposed to, and not corrupted in any way.
+ */
+
+
+#define TEST_CAPS \
+  "application/x-rtp, "                \
+  "media=(string)video, "              \
+  "clock-rate=(int)90000, "            \
+  "encoding-name=(string)H264, "       \
+  "profile-level-id=(string)4d4015, "  \
+  "payload=(int)96, "                  \
+  "ssrc=(guint)2633237432, "           \
+  "clock-base=(guint)1868267015, "     \
+  "seqnum-base=(guint)54229"
+
+
+/* RTP headers and the first 2 bytes of the payload (FU indicator and FU header)
+ */
+static const guint8 rtp_header[2][14] = {
+  {0x80, 0x60, 0xbb, 0xb7, 0x5c, 0xe9, 0x09,
+      0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x86},
+  {0x80, 0x60, 0xbb, 0xb8, 0x5c, 0xe9, 0x09,
+      0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x46}
+};
+
+static const guint rtp_header_len[] = {
+  sizeof rtp_header[0],
+  sizeof rtp_header[1]
+};
+
+static GstBuffer *header_buffer[2] = { NULL, NULL };
+
+
+/* Some payload.
+ */
+static char *payload =
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+    "0123456789ABSDEF0123456";
+
+static const guint payload_offset[] = {
+  0, 498
+};
+
+static const guint payload_len[] = {
+  498, 5
+};
+
+
+static GstBuffer *original_buffer = NULL;
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp"));
+
+
+static GstBuffer *
+_create_original_buffer (void)
+{
+  GstCaps *caps;
+
+  if (original_buffer != NULL)
+    return original_buffer;
+
+  original_buffer = gst_buffer_new ();
+  fail_unless (original_buffer != NULL);
+
+  gst_buffer_set_data (original_buffer, (guint8 *) payload, strlen (payload));
+  GST_BUFFER_TIMESTAMP (original_buffer) =
+      gst_clock_get_internal_time (gst_system_clock_obtain ());
+
+  caps = gst_caps_from_string (TEST_CAPS);
+  fail_unless (caps != NULL);
+  gst_buffer_set_caps (original_buffer, caps);
+  gst_caps_unref (caps);
+
+  return original_buffer;
+}
+
+static GstBufferList *
+_create_buffer_list (void)
+{
+  GstBufferList *list;
+  GstBufferListIterator *it;
+  GstBuffer *orig_buffer;
+  GstBuffer *buffer;
+
+  orig_buffer = _create_original_buffer ();
+  fail_if (orig_buffer == NULL);
+
+  list = gst_buffer_list_new ();
+  fail_if (list == NULL);
+
+  it = gst_buffer_list_iterate (list);
+  fail_if (it == NULL);
+
+  /*** First group, i.e. first packet. **/
+  gst_buffer_list_iterator_add_group (it);
+
+  /* Create buffer with RTP header and add it to the 1st group */
+  buffer = gst_buffer_new ();
+  GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[0], rtp_header_len[0]);
+  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+  GST_BUFFER_SIZE (buffer) = rtp_header_len[0];
+  gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
+  header_buffer[0] = buffer;
+  gst_buffer_list_iterator_add (it, buffer);
+
+  /* Create the payload buffer and add it to the 1st group
+   */
+  buffer =
+      gst_buffer_create_sub (orig_buffer, payload_offset[0], payload_len[0]);
+  fail_if (buffer == NULL);
+  gst_buffer_list_iterator_add (it, buffer);
+
+
+  /***  Second group, i.e. second packet. ***/
+
+  /* Create a new group to hold the rtp header and the payload */
+  gst_buffer_list_iterator_add_group (it);
+
+  /* Create buffer with RTP header and add it to the 2nd group */
+  buffer = gst_buffer_new ();
+  GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[1], rtp_header_len[1]);
+  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+  GST_BUFFER_SIZE (buffer) = rtp_header_len[1];
+  gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
+  header_buffer[1] = buffer;
+
+  /* Add the rtp header to the buffer list */
+  gst_buffer_list_iterator_add (it, buffer);
+
+  /* Create the payload buffer and add it to the 2d group
+   */
+  buffer =
+      gst_buffer_create_sub (orig_buffer, payload_offset[1], payload_len[1]);
+  fail_if (buffer == NULL);
+  gst_buffer_list_iterator_add (it, buffer);
+
+  gst_buffer_list_iterator_free (it);
+
+  return list;
+}
+
+
+static void
+_check_header (GstBuffer * buffer, guint index)
+{
+  guint8 *data;
+
+  fail_if (buffer == NULL);
+  fail_unless (index < 2);
+
+  fail_unless (GST_BUFFER_SIZE (buffer) == rtp_header_len[index]);
+
+  /* Can't do a memcmp() on the whole header, cause the SSRC (bytes 8-11) will
+   * most likely be changed in gstrtpbin.
+   */
+  fail_unless ((data = GST_BUFFER_DATA (buffer)) != NULL);
+  fail_unless_equals_uint64 (*(guint64 *) data, *(guint64 *) rtp_header[index]);
+  fail_unless (*(guint16 *) (data + 12) ==
+      *(guint16 *) (rtp_header[index] + 12));
+}
+
+
+static void
+_check_payload (GstBuffer * buffer, guint index)
+{
+  fail_if (buffer == NULL);
+  fail_unless (index < 2);
+
+  fail_unless (GST_BUFFER_SIZE (buffer) == payload_len[index]);
+  fail_if (GST_BUFFER_DATA (buffer) !=
+      (gpointer) (payload + payload_offset[index]));
+  fail_if (memcmp (GST_BUFFER_DATA (buffer), payload + payload_offset[index],
+          payload_len[index]));
+}
+
+
+static void
+_check_group (GstBufferListIterator * it, guint index, GstCaps * caps)
+{
+  GstBuffer *buffer;
+
+  fail_unless (it != NULL);
+  fail_unless (gst_buffer_list_iterator_n_buffers (it) == 2);
+  fail_unless (caps != NULL);
+
+  fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
+
+  fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
+      GST_BUFFER_TIMESTAMP (original_buffer));
+
+  fail_unless (gst_caps_is_equal (GST_BUFFER_CAPS (original_buffer),
+          GST_BUFFER_CAPS (buffer)));
+
+  _check_header (buffer, index);
+
+  fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
+  _check_payload (buffer, index);
+}
+
+
+static GstFlowReturn
+_sink_chain_list (GstPad * pad, GstBufferList * list)
+{
+  GstCaps *caps;
+  GstBufferListIterator *it;
+
+  caps = gst_caps_from_string (TEST_CAPS);
+  fail_unless (caps != NULL);
+
+  fail_unless (GST_IS_BUFFER_LIST (list));
+  fail_unless (gst_buffer_list_n_groups (list) == 2);
+
+  it = gst_buffer_list_iterate (list);
+  fail_if (it == NULL);
+
+  fail_unless (gst_buffer_list_iterator_next_group (it));
+  _check_group (it, 0, caps);
+
+  fail_unless (gst_buffer_list_iterator_next_group (it));
+  _check_group (it, 1, caps);
+
+  gst_caps_unref (caps);
+  gst_buffer_list_iterator_free (it);
+
+  gst_buffer_list_unref (list);
+
+  return GST_FLOW_OK;
+}
+
+
+static void
+_set_chain_functions (GstPad * pad)
+{
+  gst_pad_set_chain_list_function (pad, _sink_chain_list);
+}
+
+
+GST_START_TEST (test_bufferlist)
+{
+  GstElement *rtpbin;
+  GstPad *sinkpad;
+  GstPad *srcpad;
+  GstBufferList *list;
+
+  list = _create_buffer_list ();
+  fail_unless (list != NULL);
+
+  rtpbin = gst_check_setup_element ("gstrtpbin");
+
+  srcpad =
+      gst_check_setup_src_pad_by_name (rtpbin, &srctemplate, "send_rtp_sink_0");
+  fail_if (srcpad == NULL);
+  sinkpad =
+      gst_check_setup_sink_pad_by_name (rtpbin, &sinktemplate,
+      "send_rtp_src_0");
+  fail_if (sinkpad == NULL);
+
+  _set_chain_functions (sinkpad);
+
+  gst_pad_set_active (sinkpad, TRUE);
+  gst_element_set_state (rtpbin, GST_STATE_PLAYING);
+  fail_unless (gst_pad_push_list (srcpad, list) == GST_FLOW_OK);
+  gst_pad_set_active (sinkpad, FALSE);
+
+  gst_check_teardown_pad_by_name (rtpbin, "send_rtp_src_0");
+  gst_check_teardown_pad_by_name (rtpbin, "send_rtp_sink_0");
+  gst_check_teardown_element (rtpbin);
+}
+
+GST_END_TEST;
+
+
+
+static Suite *
+bufferlist_suite (void)
+{
+  Suite *s = suite_create ("BufferList");
+
+  TCase *tc_chain = tcase_create ("general");
+
+  /* time out after 30s. */
+  tcase_set_timeout (tc_chain, 10);
+
+  suite_add_tcase (s, tc_chain);
+  tcase_add_test (tc_chain, test_bufferlist);
+
+  return s;
+}
+
+GST_CHECK_MAIN (bufferlist);