rtpbasepayload: rtpbasedepayload: Add source-info property

Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947

17 files changed, 972 insertions, 43 deletions

diff --git a/docs/libs/gst-plugins-base-libs-docs.sgml b/docs/libs/gst-plugins-base-libs-docs.sgml
index 6b6c9966f..8149d29ab 100644
--- a/docs/libs/gst-plugins-base-libs-docs.sgml
+++ b/docs/libs/gst-plugins-base-libs-docs.sgml
@@ -110,6 +110,7 @@
         <filename>gstreamer-plugins-base-&GST_API_VERSION;.pc</filename> and adding
         <filename>-lgstrtp-&GST_API_VERSION;</filename> to the library flags.
       </para>
+      <xi:include href="xml/gstrtpmeta.xml" />
       <xi:include href="xml/gstrtpbaseaudiopayload.xml" />
       <xi:include href="xml/gstrtpbasedepayload.xml" />
       <xi:include href="xml/gstrtpbasepayload.xml" />
diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt
index 5256114ff..687c1e67d 100644
--- a/docs/libs/gst-plugins-base-libs-sections.txt
+++ b/docs/libs/gst-plugins-base-libs-sections.txt
@@ -1374,6 +1374,9 @@ GST_RTP_BASE_DEPAYLOAD_SRCPAD
 gst_rtp_base_depayload_push
 gst_rtp_base_depayload_push_list
 
+gst_rtp_base_depayload_is_source_info_enabled
+gst_rtp_base_depayload_set_source_info_enabled
+
 <SUBSECTION Standard>
 GstRTPBaseDepayloadPrivate
 GST_TYPE_RTP_BASE_DEPAYLOAD
@@ -1404,6 +1407,11 @@ gst_rtp_base_payload_push
 gst_rtp_base_payload_push_list
 gst_rtp_base_payload_set_options
 gst_rtp_base_payload_set_outcaps
+
+gst_rtp_base_payload_allocate_output_buffer
+gst_rtp_base_payload_get_source_count
+gst_rtp_base_payload_is_source_info_enabled
+gst_rtp_base_payload_set_source_info_enabled
 <SUBSECTION Standard>
 GST_TYPE_RTP_BASE_PAYLOAD
 GST_RTP_BASE_PAYLOAD
@@ -1415,6 +1423,23 @@ gst_rtp_base_payload_get_type
 </SECTION>
 
 <SECTION>
+<FILE>gstrtpmeta</FILE>
+<INCLUDE>gst/rtp/rtp.h</INCLUDE>
+GstRTPSourceMeta
+gst_buffer_add_rtp_source_meta
+gst_buffer_get_rtp_source_meta
+gst_rtp_source_meta_append_csrc
+gst_rtp_source_meta_get_info
+gst_rtp_source_meta_get_source_count
+gst_rtp_source_meta_set_ssrc
+GST_RTP_SOURCE_META_MAX_CSRC_COUNT
+<SUBSECTION Standard>
+gst_rtp_source_meta_api_get_type
+GST_RTP_SOURCE_META_API_TYPE
+GST_RTP_SOURCE_META_INFO
+</SECTION>
+
+<SECTION>
 <FILE>gstrtcpbuffer</FILE>
 <INCLUDE>gst/rtp/rtp.h</INCLUDE>
 
diff --git a/gst-libs/gst/rtp/Makefile.am b/gst-libs/gst/rtp/Makefile.am
index 4dce44894..f6b996324 100644
--- a/gst-libs/gst/rtp/Makefile.am
+++ b/gst-libs/gst/rtp/Makefile.am
@@ -10,7 +10,8 @@ libgstrtpinclude_HEADERS = \
 			   gstrtphdrext.h \
 			gstrtpbaseaudiopayload.h \
 			gstrtpbasepayload.h \
-			gstrtpbasedepayload.h
+			gstrtpbasedepayload.h \
+			gstrtpmeta.h
 
 lib_LTLIBRARIES = libgstrtp-@GST_API_VERSION@.la
 
@@ -20,7 +21,8 @@ libgstrtp_@GST_API_VERSION@_la_SOURCES = gstrtpbuffer.c \
 			   	gstrtphdrext.c \
 			        gstrtpbaseaudiopayload.c \
 				gstrtpbasepayload.c \
-				gstrtpbasedepayload.c
+				gstrtpbasedepayload.c \
+				gstrtpmeta.c
 
 built_sources = gstrtp-enumtypes.c
 built_headers = gstrtp-enumtypes.h
diff --git a/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
index 52a4c095a..378744866 100644
--- a/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
+++ b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
@@ -110,6 +110,7 @@ struct _GstRTPBaseAudioPayloadPrivate
   guint cached_max_length;
   guint cached_ptime_multiple;
   guint cached_align;
+  guint cached_csrc_count;
 
   gboolean buffer_list;
 };
@@ -453,7 +454,8 @@ gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create buffer to hold the payload */
-  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
+      payload_len, 0, 0);
 
   /* copy payload */
   gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
@@ -519,7 +521,7 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
       payload_len, GST_TIME_ARGS (timestamp));
 
   /* create just the RTP header buffer */
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
   /* set metadata */
   gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
@@ -628,7 +630,7 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
 
 
     /* create buffer to hold the payload */
-    outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+    outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
     paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
 
@@ -653,8 +655,8 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
  * mtu and min/max_ptime values. We cache those so that we don't have to redo
  * all the calculations */
 static gboolean
-gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
-    basepayload, guint * min_payload_len, guint * max_payload_len,
+gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload * basepayload,
+    guint csrc_count, guint * min_payload_len, guint * max_payload_len,
     guint * align)
 {
   GstRTPBaseAudioPayload *payload;
@@ -672,13 +674,16 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
 
   mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
 
-  /* check cached values */
+  /* check cached values. Since csrc_count may vary for each packet, we only
+   * check whether the new value exceeds the cached value and thus result in
+   * smaller payload. */
   if (G_LIKELY (priv->cached_mtu == mtu
           && priv->cached_ptime_multiple ==
           basepayload->ptime_multiple
           && priv->cached_ptime == basepayload->ptime
           && priv->cached_max_ptime == basepayload->max_ptime
-          && priv->cached_min_ptime == basepayload->min_ptime)) {
+          && priv->cached_min_ptime == basepayload->min_ptime
+          && priv->cached_csrc_count >= csrc_count)) {
     /* if nothing changed, return cached values */
     *min_payload_len = priv->cached_min_length;
     *max_payload_len = priv->cached_max_length;
@@ -697,7 +702,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
     maxptime_octets = G_MAXUINT;
   }
   /* MTU max */
-  max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
+  max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, csrc_count);
   /* round down to alignment */
   max_mtu = ALIGN_DOWN (max_mtu, *align);
 
@@ -733,6 +738,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
   priv->cached_min_length = *min_payload_len;
   priv->cached_max_length = *max_payload_len;
   priv->cached_align = *align;
+  priv->cached_csrc_count = csrc_count;
 
   return TRUE;
 }
@@ -875,8 +881,9 @@ gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
     }
   }
 
-  if (!gst_rtp_base_audio_payload_get_lengths (basepayload, &min_payload_len,
-          &max_payload_len, &align))
+  if (!gst_rtp_base_audio_payload_get_lengths (basepayload,
+          gst_rtp_base_payload_get_source_count (basepayload, buffer),
+          &min_payload_len, &max_payload_len, &align))
     goto config_error;
 
   GST_DEBUG_OBJECT (payload,
diff --git a/gst-libs/gst/rtp/gstrtpbasedepayload.c b/gst-libs/gst/rtp/gstrtpbasedepayload.c
index 9047a011b..c9c5204e1 100644
--- a/gst-libs/gst/rtp/gstrtpbasedepayload.c
+++ b/gst-libs/gst/rtp/gstrtpbasedepayload.c
@@ -30,6 +30,7 @@
 #endif
 
 #include "gstrtpbasedepayload.h"
+#include "gstrtpmeta.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
 #define GST_CAT_DEFAULT (rtpbasedepayload_debug)
@@ -57,6 +58,9 @@ struct _GstRTPBaseDepayloadPrivate
   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 */
@@ -66,10 +70,13 @@ enum
   LAST_SIGNAL
 };
 
+#define DEFAULT_SOURCE_INFO FALSE
+
 enum
 {
   PROP_0,
   PROP_STATS,
+  PROP_SOURCE_INFO,
   PROP_LAST
 };
 
@@ -183,6 +190,18 @@ gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
       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;
@@ -231,6 +250,7 @@ gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
   priv->dts = -1;
   priv->pts = -1;
   priv->duration = -1;
+  priv->source_info = DEFAULT_SOURCE_INFO;
 
   gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
 }
@@ -442,6 +462,8 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
     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);
@@ -458,6 +480,7 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
   }
 
   gst_buffer_unref (in);
+  priv->input_buffer = NULL;
 
   return ret;
 
@@ -727,6 +750,30 @@ create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
   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)
 {
@@ -759,6 +806,9 @@ set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
   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;
 }
 
@@ -959,7 +1009,15 @@ 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;
@@ -979,8 +1037,44 @@ gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
       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;
+}
diff --git a/gst-libs/gst/rtp/gstrtpbasedepayload.h b/gst-libs/gst/rtp/gstrtpbasedepayload.h
index f409ff974..42af74507 100644
--- a/gst-libs/gst/rtp/gstrtpbasedepayload.h
+++ b/gst-libs/gst/rtp/gstrtpbasedepayload.h
@@ -120,6 +120,13 @@ GstFlowReturn   gst_rtp_base_depayload_push       (GstRTPBaseDepayload *filter,
 GST_RTP_API
 GstFlowReturn   gst_rtp_base_depayload_push_list  (GstRTPBaseDepayload *filter, GstBufferList *out_list);
 
+GST_RTP_API
+gboolean        gst_rtp_base_depayload_is_source_info_enabled  (GstRTPBaseDepayload * depayload);
+
+GST_RTP_API
+void            gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
+                                                                gboolean enable);
+
 
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseDepayload, gst_object_unref)
diff --git a/gst-libs/gst/rtp/gstrtpbasepayload.c b/gst-libs/gst/rtp/gstrtpbasepayload.c
index b518ac735..23129d9ea 100644
--- a/gst-libs/gst/rtp/gstrtpbasepayload.c
+++ b/gst-libs/gst/rtp/gstrtpbasepayload.c
@@ -29,6 +29,7 @@
 #include <gst/rtp/gstrtpbuffer.h>
 
 #include "gstrtpbasepayload.h"
+#include "gstrtpmeta.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug);
 #define GST_CAT_DEFAULT (rtpbasepayload_debug)
@@ -44,6 +45,9 @@ struct _GstRTPBasePayloadPrivate
 
   gboolean pt_set;
 
+  gboolean source_info;
+  GstBuffer *input_meta_buffer;
+
   guint64 base_offset;
   gint64 base_rtime;
   guint64 base_rtime_hz;
@@ -84,6 +88,7 @@ enum
 #define DEFAULT_PERFECT_RTPTIME         TRUE
 #define DEFAULT_PTIME_MULTIPLE          0
 #define DEFAULT_RUNNING_TIME            GST_CLOCK_TIME_NONE
+#define DEFAULT_SOURCE_INFO             FALSE
 
 enum
 {
@@ -100,6 +105,7 @@ enum
   PROP_PERFECT_RTPTIME,
   PROP_PTIME_MULTIPLE,
   PROP_STATS,
+  PROP_SOURCE_INFO,
   PROP_LAST
 };
 
@@ -299,6 +305,19 @@ gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
       g_param_spec_boxed ("stats", "Statistics", "Various statistics",
           GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 
+  /**
+   * GstRTPBasePayload:source-info:
+   *
+   * Enable writing the CSRC field in allocated RTP header based on RTP source
+   * information found in the input buffer's #GstRTPSourceMeta.
+   *
+   * Since: 1.16
+   **/
+  g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
+      g_param_spec_boolean ("source-info", "RTP source information",
+          "Write CSRC based on buffer meta RTP source information",
+          DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
+
   gstelement_class->change_state = gst_rtp_base_payload_change_state;
 
   klass->get_caps = gst_rtp_base_payload_getcaps_default;
@@ -351,6 +370,7 @@ gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class)
   priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
   priv->ssrc_random = (rtpbasepayload->ssrc == -1);
   priv->pt_set = FALSE;
+  priv->source_info = DEFAULT_SOURCE_INFO;
 
   rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
   rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
@@ -633,6 +653,15 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
   if (!rtpbasepayload->priv->negotiated)
     goto not_negotiated;
 
+  if (rtpbasepayload->priv->source_info) {
+    /* Save a copy of meta (instead of taking an extra reference before
+     * handle_buffer) to make the meta available when allocating a output
+     * buffer. */
+    rtpbasepayload->priv->input_meta_buffer = gst_buffer_new ();
+    gst_buffer_copy_into (rtpbasepayload->priv->input_meta_buffer, buffer,
+        GST_BUFFER_COPY_META, 0, -1);
+  }
+
   if (gst_pad_check_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
     if (!gst_rtp_base_payload_negotiate (rtpbasepayload)) {
       gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload));
@@ -646,6 +675,8 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
 
   ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
 
+  gst_buffer_replace (&rtpbasepayload->priv->input_meta_buffer, NULL);
+
   return ret;
 
   /* ERRORS */
@@ -1156,6 +1187,25 @@ map_failed:
   }
 }
 
+static gboolean
+foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
+{
+  GType drop_api_type = (GType) GPOINTER_TO_INT (user_data);
+  const GstMetaInfo *info = (*meta)->info;
+
+  if (info->api == drop_api_type)
+    *meta = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+filter_meta (GstBuffer ** buffer, guint idx, gpointer user_data)
+{
+  return gst_buffer_foreach_meta (*buffer, foreach_metadata_drop,
+      GINT_TO_POINTER (GST_RTP_SOURCE_META_API_TYPE));
+}
+
 /* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
  * before the buffer is pushed. */
 static GstFlowReturn
@@ -1252,11 +1302,14 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
   }
 
   /* set ssrc, payload type, seq number, caps and rtptime */
+  /* remove unwanted meta */
   if (is_list) {
     gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), set_headers, &data);
+    gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), filter_meta, NULL);
   } else {
     GstBuffer *buf = GST_BUFFER_CAST (obj);
     set_headers (&buf, 0, &data);
+    filter_meta (&buf, 0, NULL);
   }
 
   priv->next_seqnum = data.seqnum;
@@ -1267,8 +1320,8 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
       (is_list) ? -1 : gst_buffer_get_size (GST_BUFFER (obj)),
       payload->seqnum, data.rtptime, GST_TIME_ARGS (data.pts));
 
-  if (g_atomic_int_compare_and_exchange (&payload->
-          priv->notified_first_timestamp, 1, 0)) {
+  if (g_atomic_int_compare_and_exchange (&payload->priv->
+          notified_first_timestamp, 1, 0)) {
     g_object_notify (G_OBJECT (payload), "timestamp");
     g_object_notify (G_OBJECT (payload), "seqnum");
   }
@@ -1351,6 +1404,62 @@ gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
   return res;
 }
 
+/**
+ * gst_rtp_base_payload_allocate_output_buffer:
+ * @payload: a #GstRTPBasePayload
+ * @payload_len: the length of the payload
+ * @pad_len: the amount of padding
+ * @csrc_count: the minimum number of CSRC entries
+ *
+ * Allocate a new #GstBuffer with enough data to hold an RTP packet with
+ * minimum @csrc_count CSRCs, a payload length of @payload_len and padding of
+ * @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional
+ * CSRCs may be allocated and filled with RTP source information.
+ *
+ * Returns: A newly allocated buffer that can hold an RTP packet with given
+ * parameters.
+ *
+ * Since: 1.16
+ */
+GstBuffer *
+gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
+    guint payload_len, guint8 pad_len, guint8 csrc_count)
+{
+  GstBuffer *buffer = NULL;
+
+  if (payload->priv->input_meta_buffer != NULL) {
+    GstRTPSourceMeta *meta =
+        gst_buffer_get_rtp_source_meta (payload->priv->input_meta_buffer);
+    if (meta != NULL) {
+      guint total_csrc_count, idx, i;
+      GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+      total_csrc_count = csrc_count + meta->csrc_count +
+          (meta->ssrc_valid ? 1 : 0);
+      total_csrc_count = MIN (total_csrc_count, 15);
+      buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len,
+          total_csrc_count);
+
+      gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp);
+
+      /* Skip CSRC fields requested by derived class and fill CSRCs from meta.
+       * Finally append the SSRC as a new CSRC. */
+      idx = csrc_count;
+      for (i = 0; i < meta->csrc_count && idx < 15; i++, idx++)
+        gst_rtp_buffer_set_csrc (&rtp, idx, meta->csrc[i]);
+      if (meta->ssrc_valid && idx < 15)
+        gst_rtp_buffer_set_csrc (&rtp, idx, meta->ssrc);
+
+      gst_rtp_buffer_unmap (&rtp);
+    }
+  }
+
+  if (buffer == NULL)
+    buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len, csrc_count);
+
+  return buffer;
+}
+
 static GstStructure *
 gst_rtp_base_payload_create_stats (GstRTPBasePayload * rtpbasepayload)
 {
@@ -1421,6 +1530,10 @@ gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
     case PROP_PTIME_MULTIPLE:
       rtpbasepayload->ptime_multiple = g_value_get_int64 (value);
       break;
+    case PROP_SOURCE_INFO:
+      gst_rtp_base_payload_set_source_info_enabled (rtpbasepayload,
+          g_value_get_boolean (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1484,6 +1597,10 @@ gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
       g_value_take_boxed (value,
           gst_rtp_base_payload_create_stats (rtpbasepayload));
       break;
+    case PROP_SOURCE_INFO:
+      g_value_set_boolean (value,
+          gst_rtp_base_payload_is_source_info_enabled (rtpbasepayload));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1551,3 +1668,68 @@ gst_rtp_base_payload_change_state (GstElement * element,
   }
   return ret;
 }
+
+/**
+ * gst_rtp_base_payload_set_source_info_enabled:
+ * @payload: a #GstRTPBasePayload
+ * @enable: whether to add contributing sources to RTP packets
+ *
+ * Enable or disable adding contributing sources to RTP packets from
+ * #GstRTPSourceMeta.
+ *
+ * Since: 1.16
+ **/
+void
+gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
+    gboolean enable)
+{
+  payload->priv->source_info = enable;
+}
+
+/**
+ * gst_rtp_base_payload_is_source_info_enabled:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Queries whether the payloader will add contributing sources (CSRCs) to the
+ * RTP header from #GstRTPSourceMeta.
+ *
+ * Returns: %TRUE if source-info is enabled.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)
+{
+  return payload->priv->source_info;
+}
+
+
+/**
+ * gst_rtp_base_payload_get_source_count:
+ * @payload: a #GstRTPBasePayload
+ * @buffer: (transfer none): a #GstBuffer, typically the buffer to payload
+ *
+ * Count the total number of RTP sources found in the meta of @buffer, which
+ * will be automically added by gst_rtp_base_payload_allocate_output_buffer().
+ * If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
+ *
+ * Returns: The number of sources.
+ *
+ * Since: 1.16
+ **/
+guint
+gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
+    GstBuffer * buffer)
+{
+  guint count = 0;
+
+  g_return_val_if_fail (buffer != NULL, 0);
+
+  if (gst_rtp_base_payload_is_source_info_enabled (payload)) {
+    GstRTPSourceMeta *meta = gst_buffer_get_rtp_source_meta (buffer);
+    if (meta != NULL)
+      count = gst_rtp_source_meta_get_source_count (meta);
+  }
+
+  return count;
+}
diff --git a/gst-libs/gst/rtp/gstrtpbasepayload.h b/gst-libs/gst/rtp/gstrtpbasepayload.h
index b477e8a21..8b9e984ae 100644
--- a/gst-libs/gst/rtp/gstrtpbasepayload.h
+++ b/gst-libs/gst/rtp/gstrtpbasepayload.h
@@ -172,6 +172,22 @@ GST_RTP_API
 GstFlowReturn   gst_rtp_base_payload_push_list          (GstRTPBasePayload *payload,
                                                          GstBufferList *list);
 
+GST_RTP_API
+GstBuffer *     gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
+                                                             guint payload_len, guint8 pad_len,
+                                                             guint8 csrc_count);
+
+GST_RTP_API
+void            gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
+                                                              gboolean enable);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload);
+
+GST_RTP_API
+guint           gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
+                                                       GstBuffer * buffer);
+
 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBasePayload, gst_object_unref)
 #endif
diff --git a/gst-libs/gst/rtp/gstrtpmeta.c b/gst-libs/gst/rtp/gstrtpmeta.c
new file mode 100644
index 000000000..9028db82d
--- /dev/null
+++ b/gst-libs/gst/rtp/gstrtpmeta.c
@@ -0,0 +1,229 @@
+/* GStreamer
+ * Copyright (C) <2016> Stian Selnes <stian@pexip.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 "gstrtpmeta.h"
+#include <string.h>
+
+/**
+ * SECTION:gstrtpmeta
+ * @title: GstMeta for RTP
+ * @short_description: RTP related GstMeta
+ *
+ */
+
+/**
+ * gst_buffer_add_rtp_source_meta:
+ * @buffer: a #GstBuffer
+ * @ssrc: (allow-none) (transfer none): pointer to the SSRC
+ * @csrc: (allow-none) (transfer none): pointer to the CSRCs
+ * @csrc_count: number of elements in @csrc
+ *
+ * Attaches RTP source information to @buffer.
+ *
+ * Returns: (transfer none): the #GstRTPSourceMeta on @buffer.
+ *
+ * Since: 1.16
+ */
+GstRTPSourceMeta *
+gst_buffer_add_rtp_source_meta (GstBuffer * buffer, const guint32 * ssrc,
+    const guint * csrc, guint csrc_count)
+{
+  gint i;
+  GstRTPSourceMeta *meta;
+
+  g_return_val_if_fail (buffer != NULL, NULL);
+  g_return_val_if_fail (csrc_count <= GST_RTP_SOURCE_META_MAX_CSRC_COUNT, NULL);
+  g_return_val_if_fail (csrc_count == 0 || csrc != NULL, NULL);
+
+  meta = (GstRTPSourceMeta *) gst_buffer_add_meta (buffer,
+      GST_RTP_SOURCE_META_INFO, NULL);
+  if (!meta)
+    return NULL;
+
+  if (ssrc != NULL) {
+    meta->ssrc = *ssrc;
+    meta->ssrc_valid = TRUE;
+  } else {
+    meta->ssrc_valid = FALSE;
+  }
+
+  meta->csrc_count = csrc_count;
+  for (i = 0; i < csrc_count; i++) {
+    meta->csrc[i] = csrc[i];
+  }
+
+  return meta;
+}
+
+/**
+ * gst_buffer_get_rtp_source_meta:
+ * @buffer: a #GstBuffer
+ *
+ * Find the #GstRTPSourceMeta on @buffer.
+ *
+ * Returns: (transfer none): the #GstRTPSourceMeta or %NULL when there
+ * is no such metadata on @buffer.
+ *
+ * Since: 1.16
+ */
+GstRTPSourceMeta *
+gst_buffer_get_rtp_source_meta (GstBuffer * buffer)
+{
+  return (GstRTPSourceMeta *) gst_buffer_get_meta (buffer,
+      gst_rtp_source_meta_api_get_type ());
+}
+
+static gboolean
+gst_rtp_source_meta_transform (GstBuffer * dst, GstMeta * meta,
+    GstBuffer * src, GQuark type, gpointer data)
+{
+  if (GST_META_TRANSFORM_IS_COPY (type)) {
+    GstRTPSourceMeta *smeta = (GstRTPSourceMeta *) meta;
+    GstRTPSourceMeta *dmeta;
+    guint32 *ssrc = smeta->ssrc_valid ? &smeta->ssrc : NULL;
+
+    dmeta = gst_buffer_add_rtp_source_meta (dst, ssrc, smeta->csrc,
+        smeta->csrc_count);
+    if (dmeta == NULL)
+      return FALSE;
+  } else {
+    /* return FALSE, if transform type is not supported */
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+/**
+ * gst_rtp_source_meta_get_source_count:
+ * @meta: a #GstRTPSourceMeta
+ *
+ * Count the total number of RTP sources found in @meta, both SSRC and CSRC.
+ *
+ * Returns: The number of RTP sources
+ *
+ * Since: 1.16
+ */
+guint
+gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta)
+{
+  /* Never return more than a count of 15 so that the returned value
+   * conveniently can be used as argument 'csrc_count' in
+   * gst_rtp_buffer-functions. */
+  guint ssrc_count = meta->ssrc_valid ? 1 : 0;
+  return MIN (meta->csrc_count + ssrc_count, 15);
+}
+
+/**
+ * gst_rtp_source_meta_set_ssrc:
+ * @meta: a #GstRTPSourceMeta
+ * @ssrc: (allow-none) (transfer none): pointer to the SSRC
+ *
+ * Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset.
+ *
+ * Returns: %TRUE on success, %FALSE otherwise.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_source_meta_set_ssrc (GstRTPSourceMeta * meta, guint32 * ssrc)
+{
+  if (ssrc != NULL) {
+    meta->ssrc = *ssrc;
+    meta->ssrc_valid = TRUE;
+  } else {
+    meta->ssrc_valid = FALSE;
+  }
+
+  return TRUE;
+}
+
+/**
+ * gst_rtp_source_meta_append_csrc:
+ * @meta: a #GstRTPSourceMeta
+ * @csrc: the csrcs to append
+ * @csrc_count: number of elements in @csrc
+ *
+ * Appends @csrc to the list of contributing sources in @meta.
+ *
+ * Returns: %TRUE if all elements in @csrc was added, %FALSE otherwise.
+ *
+ * Since: 1.16
+ **/
+gboolean
+gst_rtp_source_meta_append_csrc (GstRTPSourceMeta * meta, const guint32 * csrc,
+    guint csrc_count)
+{
+  gint i;
+  guint new_csrc_count = meta->csrc_count + csrc_count;
+
+  if (new_csrc_count > GST_RTP_SOURCE_META_MAX_CSRC_COUNT)
+    return FALSE;
+
+  for (i = 0; i < csrc_count; i++)
+    meta->csrc[meta->csrc_count + i] = csrc[i];
+  meta->csrc_count = new_csrc_count;
+
+  return TRUE;
+}
+
+GType
+gst_rtp_source_meta_api_get_type (void)
+{
+  static volatile GType type = 0;
+  static const gchar *tags[] = { NULL };
+
+  if (g_once_init_enter (&type)) {
+    GType _type = gst_meta_api_type_register ("GstRTPSourceMetaAPI", tags);
+    g_once_init_leave (&type, _type);
+  }
+  return type;
+}
+
+static gboolean
+gst_rtp_source_meta_init (GstMeta * meta, gpointer params, GstBuffer * buffer)
+{
+  GstRTPSourceMeta *dmeta = (GstRTPSourceMeta *) meta;
+
+  dmeta->ssrc_valid = FALSE;
+  dmeta->csrc_count = 0;
+
+  return TRUE;
+}
+
+const GstMetaInfo *
+gst_rtp_source_meta_get_info (void)
+{
+  static const GstMetaInfo *rtp_source_meta_info = NULL;
+
+  if (g_once_init_enter (&rtp_source_meta_info)) {
+    const GstMetaInfo *meta = gst_meta_register (GST_RTP_SOURCE_META_API_TYPE,
+        "GstRTPSourceMeta",
+        sizeof (GstRTPSourceMeta),
+        gst_rtp_source_meta_init,
+        (GstMetaFreeFunction) NULL,
+        gst_rtp_source_meta_transform);
+    g_once_init_leave (&rtp_source_meta_info, meta);
+  }
+  return rtp_source_meta_info;
+}
diff --git a/gst-libs/gst/rtp/gstrtpmeta.h b/gst-libs/gst/rtp/gstrtpmeta.h
new file mode 100644
index 000000000..956cff7ef
--- /dev/null
+++ b/gst-libs/gst/rtp/gstrtpmeta.h
@@ -0,0 +1,79 @@
+/* GStreamer
+ * Copyright (C) <2016> Stian Selnes <stian@pexip.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.
+ */
+
+#ifndef __GST_RTP_META_H__
+#define __GST_RTP_META_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+#define GST_RTP_SOURCE_META_API_TYPE  (gst_rtp_source_meta_api_get_type())
+#define GST_RTP_SOURCE_META_INFO  (gst_rtp_source_meta_get_info())
+typedef struct _GstRTPSourceMeta GstRTPSourceMeta;
+
+#define GST_RTP_SOURCE_META_MAX_CSRC_COUNT 15
+
+/**
+ * GstRTPSourceMeta:
+ * @meta: parent #GstMeta
+ * @ssrc: the SSRC
+ * @ssrc_valid: whether @ssrc is set and valid
+ * @csrc: (allow-none): pointer to the CSRCs
+ * @csrc_count: number of elements in @csrc
+ *
+ * Meta describing the source(s) of the buffer.
+ *
+ * Since: 1.16
+ */
+struct _GstRTPSourceMeta
+{
+  GstMeta meta;
+
+  guint32 ssrc;
+  gboolean ssrc_valid;
+  guint32 csrc[GST_RTP_SOURCE_META_MAX_CSRC_COUNT];
+  guint csrc_count;
+};
+
+GST_RTP_API
+GType               gst_rtp_source_meta_api_get_type     (void);
+
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_add_rtp_source_meta       (GstBuffer * buf, const guint32 * ssrc,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_get_rtp_source_meta       (GstBuffer * buf);
+
+GST_RTP_API
+guint               gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_set_ssrc         (GstRTPSourceMeta * meta, guint32 * ssrc);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_append_csrc      (GstRTPSourceMeta * meta,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+const GstMetaInfo * gst_rtp_source_meta_get_info         (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_META_H__ */
diff --git a/gst-libs/gst/rtp/meson.build b/gst-libs/gst/rtp/meson.build
index 25d3900dd..f47ec6592 100644
--- a/gst-libs/gst/rtp/meson.build
+++ b/gst-libs/gst/rtp/meson.build
@@ -3,6 +3,7 @@ rtp_sources = [
   'gstrtcpbuffer.c',
   'gstrtppayloads.c',
   'gstrtphdrext.c',
+  'gstrtpmeta.c',
   'gstrtpbaseaudiopayload.c',
   'gstrtpbasepayload.c',
   'gstrtpbasedepayload.c'
@@ -16,6 +17,7 @@ rtp_headers = [
   'gstrtpbuffer.h',
   'gstrtpdefs.h',
   'gstrtphdrext.h',
+  'gstrtpmeta.h',
   'gstrtppayloads.h',
   'rtp-prelude.h',
   'rtp.h',
diff --git a/gst-libs/gst/rtp/rtp.h b/gst-libs/gst/rtp/rtp.h
index 546d4aef3..0e6633bd8 100644
--- a/gst-libs/gst/rtp/rtp.h
+++ b/gst-libs/gst/rtp/rtp.h
@@ -30,6 +30,7 @@
 #include <gst/rtp/gstrtpbaseaudiopayload.h>
 #include <gst/rtp/gstrtpbasepayload.h>
 #include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/rtp/gstrtpmeta.h>
 #include <gst/rtp/gstrtp-enumtypes.h>
 
 #endif /* __GST_RTP_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index f1569a6b6..6223650e9 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -245,6 +245,7 @@ check_PROGRAMS = \
 	libs/rtp \
 	libs/rtpbasedepayload \
 	libs/rtpbasepayload \
+	libs/rtpmeta \
 	libs/rtsp \
 	libs/rtspconnection \
 	libs/sdp \
@@ -581,6 +582,13 @@ libs_rtpbasedepayload_LDADD = \
 	$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
 	$(GST_BASE_LIBS) $(LDADD)
 
+libs_rtpmeta_CFLAGS = \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(AM_CFLAGS)
+libs_rtpmeta_LDADD = \
+	$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
+	$(GST_BASE_LIBS) $(LDADD)
+
 libs_rtsp_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
 	$(AM_CFLAGS)
diff --git a/tests/check/libs/.gitignore b/tests/check/libs/.gitignore
index 64a1eb63e..78cf13f11 100644
--- a/tests/check/libs/.gitignore
+++ b/tests/check/libs/.gitignore
@@ -29,6 +29,7 @@ profile
 rtp
 rtpbasedepayload
 rtpbasepayload
+rtpmeta
 rtsp
 rtspconnection
 sdp
diff --git a/tests/check/libs/rtpbasedepayload.c b/tests/check/libs/rtpbasedepayload.c
index f24b0c46c..01f1a372d 100644
--- a/tests/check/libs/rtpbasedepayload.c
+++ b/tests/check/libs/rtpbasedepayload.c
@@ -23,8 +23,8 @@
 
 #include <gst/gst.h>
 #include <gst/check/gstcheck.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/check/gstharness.h>
+#include <gst/rtp/rtp.h>
 
 #define DEFAULT_CLOCK_RATE (42)
 
@@ -317,22 +317,14 @@ validate_event (guint index, const gchar * name, const gchar * field, ...)
   va_end (var_args);
 }
 
-#define push_rtp_buffer(state, field, ...) \
-	push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
-#define push_rtp_buffer_fails(state, error, field, ...) \
-        push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
-
 static void
-push_rtp_buffer_full (State * state, GstFlowReturn expected,
-    const gchar * field, ...)
+rtp_buffer_set_valist (GstBuffer * buf, const gchar * field, va_list var_args,
+    gboolean * extra_ref_)
 {
-  GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
   GstRTPBuffer rtp = { NULL };
   gboolean mapped = FALSE;
   gboolean extra_ref = FALSE;
-  va_list var_args;
 
-  va_start (var_args, field);
   while (field) {
     if (!g_strcmp0 (field, "pts")) {
       GstClockTime pts = va_arg (var_args, GstClockTime);
@@ -366,13 +358,18 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
         gst_rtp_buffer_set_ssrc (&rtp, ssrc);
       } else if (!g_strcmp0 (field, "extra-ref")) {
         extra_ref = va_arg (var_args, gboolean);
+        if (extra_ref_)
+          *extra_ref_ = extra_ref;
+      } else if (!g_strcmp0 (field, "csrc")) {
+        guint idx = va_arg (var_args, guint);
+        guint csrc = va_arg (var_args, guint);
+        gst_rtp_buffer_set_csrc (&rtp, idx, csrc);
       } else {
         fail ("test cannot set unknown buffer field '%s'", field);
       }
     }
     field = va_arg (var_args, const gchar *);
   }
-  va_end (var_args);
 
   if (mapped) {
     gst_rtp_buffer_unmap (&rtp);
@@ -380,6 +377,34 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
 
   if (extra_ref)
     gst_buffer_ref (buf);
+}
+
+static void
+rtp_buffer_set (GstBuffer * buf, const gchar * field, ...)
+{
+  va_list var_args;
+
+  va_start (var_args, field);
+  rtp_buffer_set_valist (buf, field, var_args, NULL);
+  va_end (var_args);
+}
+
+#define push_rtp_buffer(state, field, ...) \
+    push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
+#define push_rtp_buffer_fails(state, error, field, ...) \
+    push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
+
+static void
+push_rtp_buffer_full (State * state, GstFlowReturn expected,
+    const gchar * field, ...)
+{
+  GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
+  va_list var_args;
+  gboolean extra_ref = FALSE;
+
+  va_start (var_args, field);
+  rtp_buffer_set_valist (buf, field, var_args, &extra_ref);
+  va_end (var_args);
 
   fail_unless_equals_int (gst_pad_push (state->srcpad, buf), expected);
 
@@ -388,7 +413,7 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
 }
 
 #define push_buffer(state, field, ...) \
-	push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
+    push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
 
 static void
 push_buffer_full (State * state, GstFlowReturn expected,
@@ -1237,7 +1262,64 @@ GST_START_TEST (rtp_base_depayload_clock_base_test)
   destroy_depayloader (state);
 }
 
-GST_END_TEST static Suite *
+GST_END_TEST
+/* basedepayloader has a property source-info that will add
+ * GstRTPSourceMeta to the output buffer with RTP source information, such as
+ * SSRC and CSRCs. The is useful for letting downstream know about the origin
+ * of the stream. */
+GST_START_TEST (rtp_base_depayload_source_info_test)
+{
+  GstHarness *h;
+  GstRtpDummyDepay *depay;
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint seq = 0;
+
+  depay = rtp_dummy_depay_new ();
+  h = gst_harness_new_with_element (GST_ELEMENT_CAST (depay), "sink", "src");
+  gst_harness_set_src_caps_str (h, "application/x-rtp");
+
+  /* Property enabled should always add meta, also when there is only SSRC and
+   * no CSRC. */
+  g_object_set (depay, "source-info", TRUE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, 0x11);
+  fail_unless_equals_int (meta->csrc_count, 0);
+  gst_buffer_unref (buffer);
+
+  /* Both SSRC and CSRC should be added to the meta */
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 2);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, "csrc", 0, 0x22,
+      "csrc", 1, 0x33, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, 0x11);
+  fail_unless_equals_int (meta->csrc_count, 2);
+  fail_unless_equals_int (meta->csrc[0], 0x22);
+  fail_unless_equals_int (meta->csrc[1], 0x33);
+  gst_buffer_unref (buffer);
+
+  /* Property disabled should never add meta */
+  g_object_set (depay, "source-info", FALSE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  g_object_unref (depay);
+  gst_harness_teardown (h);
+}
+
+GST_END_TEST;
+
+
+static Suite *
 rtp_basepayloading_suite (void)
 {
   Suite *s = suite_create ("rtp_base_depayloading_test");
@@ -1265,6 +1347,8 @@ rtp_basepayloading_suite (void)
   tcase_add_test (tc_chain, rtp_base_depayload_play_speed_test);
   tcase_add_test (tc_chain, rtp_base_depayload_clock_base_test);
 
+  tcase_add_test (tc_chain, rtp_base_depayload_source_info_test);
+
   return s;
 }
 
diff --git a/tests/check/libs/rtpbasepayload.c b/tests/check/libs/rtpbasepayload.c
index 9385ce949..9bda37407 100644
--- a/tests/check/libs/rtpbasepayload.c
+++ b/tests/check/libs/rtpbasepayload.c
@@ -23,8 +23,8 @@
 
 #include <gst/gst.h>
 #include <gst/check/gstcheck.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtpbasepayload.h>
+#include <gst/check/gstharness.h>
+#include <gst/rtp/rtp.h>
 
 #define DEFAULT_CLOCK_RATE (42)
 #define BUFFER_BEFORE_LIST (10)
@@ -121,7 +121,9 @@ gst_rtp_dummy_pay_handle_buffer (GstRTPBasePayload * pay, GstBuffer * buffer)
     }
   }
 
-  paybuffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  paybuffer =
+      gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
+      0, 0, 0);
 
   GST_BUFFER_PTS (paybuffer) = GST_BUFFER_PTS (buffer);
   GST_BUFFER_OFFSET (paybuffer) = GST_BUFFER_OFFSET (buffer);
@@ -444,20 +446,11 @@ validate_buffers_received (guint received_buffers)
 }
 
 static void
-validate_buffer (guint index, const gchar * field, ...)
+validate_buffer_valist (GstBuffer * buf, const gchar * field, va_list var_args)
 {
-  GstBuffer *buf;
   GstRTPBuffer rtp = { NULL };
   gboolean mapped = FALSE;
-  va_list var_args;
-
-  fail_if (index >= g_list_length (buffers));
-  buf = GST_BUFFER (g_list_nth_data (buffers, index));
-  fail_if (buf == NULL);
 
-  GST_TRACE ("%" GST_PTR_FORMAT, buf);
-
-  va_start (var_args, field);
   while (field) {
     if (!g_strcmp0 (field, "pts")) {
       GstClockTime pts = va_arg (var_args, GstClockTime);
@@ -489,13 +482,20 @@ validate_buffer (guint index, const gchar * field, ...)
       } else if (!g_strcmp0 (field, "ssrc")) {
         guint32 ssrc = va_arg (var_args, guint);
         fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
+      } else if (!g_strcmp0 (field, "csrc")) {
+        guint idx = va_arg (var_args, guint);
+        guint csrc = va_arg (var_args, guint);
+        fail_unless_equals_int (gst_rtp_buffer_get_csrc (&rtp, idx), csrc);
+      } else if (!g_strcmp0 (field, "csrc-count")) {
+        guint csrc_count = va_arg (var_args, guint);
+        fail_unless_equals_int (gst_rtp_buffer_get_csrc_count (&rtp),
+            csrc_count);
       } else {
         fail ("test cannot validate unknown buffer field '%s'", field);
       }
     }
     field = va_arg (var_args, const gchar *);
   }
-  va_end (var_args);
 
   if (mapped) {
     gst_rtp_buffer_unmap (&rtp);
@@ -503,6 +503,33 @@ validate_buffer (guint index, const gchar * field, ...)
 }
 
 static void
+validate_buffer1 (GstBuffer * buf, const gchar * field, ...)
+{
+  va_list var_args;
+
+  va_start (var_args, field);
+  validate_buffer_valist (buf, field, var_args);
+  va_end (var_args);
+}
+
+static void
+validate_buffer (guint index, const gchar * field, ...)
+{
+  GstBuffer *buf;
+  va_list var_args;
+
+  fail_if (index >= g_list_length (buffers));
+  buf = GST_BUFFER (g_list_nth_data (buffers, index));
+  fail_if (buf == NULL);
+
+  GST_TRACE ("%" GST_PTR_FORMAT, buf);
+
+  va_start (var_args, field);
+  validate_buffer_valist (buf, field, var_args);
+  va_end (var_args);
+}
+
+static void
 get_buffer_field (guint index, const gchar * field, ...)
 {
   GstBuffer *buf;
@@ -1775,6 +1802,59 @@ GST_START_TEST (rtp_base_payload_property_stats_test)
 
 GST_END_TEST;
 
+/* basepayloader has a property source-info that makes it aware of RTP
+ * source information passed as GstRTPSourceMeta on the input buffers. All
+ * sources found in the meta will be added to the list of CSRCs in the RTP
+ * header. A useful scenario for this is, for instance, to signal which
+ * sources contributed to a mixed audio stream. */
+GST_START_TEST (rtp_base_payload_property_source_info_test)
+{
+  GstHarness *h;
+  GstRtpDummyPay *pay;
+  GstBuffer *buffer;
+  guint csrc_count = 2;
+  const guint32 csrc[] = { 0x11, 0x22 };
+  const guint32 ssrc = 0x33;
+
+  pay = rtp_dummy_pay_new ();
+  h = gst_harness_new_with_element (GST_ELEMENT_CAST (pay), "sink", "src");
+  gst_harness_set_src_caps_str (h, "application/x-rtp");
+
+  /* Input buffer has no meta, payloader should not add CSRC */
+  g_object_set (pay, "source-info", TRUE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  buffer = gst_harness_push_and_pull (h, buffer);
+  validate_buffer1 (buffer, "csrc-count", 0, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  /* Input buffer has meta, payloader should add CSRC */
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  fail_unless (gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc,
+          csrc_count));
+  buffer = gst_harness_push_and_pull (h, buffer);
+  /* The meta SSRC should be added as the last contributing source */
+  validate_buffer1 (buffer, "csrc-count", 3, "csrc", 0, csrc[0],
+      "csrc", 1, csrc[1], "csrc", 2, ssrc, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  /* When property is disabled, the meta should be ignored and no CSRC
+   * added. */
+  g_object_set (pay, "source-info", FALSE, NULL);
+  buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
+  fail_unless (gst_buffer_add_rtp_source_meta (buffer, NULL, csrc, csrc_count));
+  buffer = gst_harness_push_and_pull (h, buffer);
+  validate_buffer1 (buffer, "csrc-count", 0, NULL);
+  fail_if (gst_buffer_get_rtp_source_meta (buffer));
+  gst_buffer_unref (buffer);
+
+  g_object_unref (pay);
+  gst_harness_teardown (h);
+}
+
+GST_END_TEST;
+
 /* push a single buffer to the payloader which should successfully payload it
  * into an RTP packet. besides the payloaded RTP packet there should be the
  * three events initial events: stream-start, caps and segment. because of that
@@ -1878,6 +1958,7 @@ rtp_basepayloading_suite (void)
   tcase_add_test (tc_chain, rtp_base_payload_property_perfect_rtptime_test);
   tcase_add_test (tc_chain, rtp_base_payload_property_ptime_multiple_test);
   tcase_add_test (tc_chain, rtp_base_payload_property_stats_test);
+  tcase_add_test (tc_chain, rtp_base_payload_property_source_info_test);
 
   tcase_add_test (tc_chain, rtp_base_payload_framerate_attribute);
   tcase_add_test (tc_chain, rtp_base_payload_max_framerate_attribute);
diff --git a/tests/check/libs/rtpmeta.c b/tests/check/libs/rtpmeta.c
new file mode 100644
index 000000000..312f3916e
--- /dev/null
+++ b/tests/check/libs/rtpmeta.c
@@ -0,0 +1,110 @@
+/* GStreamer RTP meta unit tests
+ * Copyright (C) 2016  Stian Selnes <stian@pexip.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.
+ */
+
+#include <gst/gst.h>
+#include <gst/check/gstcheck.h>
+#include <gst/rtp/rtp.h>
+
+GST_START_TEST (test_rtp_source_meta_set_get_sources)
+{
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint32 ssrc = 1000, ssrc2 = 2000;
+  const guint32 csrc[] = {
+    0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14
+  };
+
+  buffer = gst_buffer_new ();
+  meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 12);
+
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc);
+  for (gint i = 0; i < 12; i++)
+    fail_unless_equals_int (meta->csrc[i], csrc[i]);
+
+  /* Unset the ssrc */
+  fail_unless (gst_rtp_source_meta_set_ssrc (meta, NULL));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12);
+  fail_if (meta->ssrc_valid);
+
+  /* Set the ssrc again */
+  fail_unless (gst_rtp_source_meta_set_ssrc (meta, &ssrc2));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc2);
+
+  /* Append multiple csrcs */
+  fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[12], 2));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
+  for (gint i = 0; i < 14; i++)
+    fail_unless_equals_int (meta->csrc[i], csrc[i]);
+
+  gst_buffer_unref (buffer);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtp_source_meta_set_get_max_sources)
+{
+  GstBuffer *buffer;
+  GstRTPSourceMeta *meta;
+  guint32 ssrc = 1000;
+  const guint32 csrc[16] = { 0, };
+
+  buffer = gst_buffer_new ();
+  meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 14);
+
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
+  fail_unless_equals_int (meta->csrc_count, 14);
+  fail_unless (meta->ssrc_valid);
+  fail_unless_equals_int (meta->ssrc, ssrc);
+
+  /* Append one more csrc */
+  /* The source count should cap at 15 for convenient use with
+   * gst_rtp_buffer-functions! */
+  fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[14], 1));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
+  fail_unless_equals_int (meta->csrc_count, 15);
+
+  /* Try to append one more csrc, but we've reached max */
+  fail_if (gst_rtp_source_meta_append_csrc (meta, &csrc[15], 1));
+  fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
+  fail_unless_equals_int (meta->csrc_count, 15);
+
+  gst_buffer_unref (buffer);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtp_meta_suite (void)
+{
+  Suite *s = suite_create ("rtp_meta_tests");
+  TCase *tc_chain;
+
+  suite_add_tcase (s, (tc_chain = tcase_create ("GstRTPSourceMeta")));
+  tcase_add_test (tc_chain, test_rtp_source_meta_set_get_sources);
+  tcase_add_test (tc_chain, test_rtp_source_meta_set_get_max_sources);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtp_meta)