basertppay: allow subclasses to influence RTP time

Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.

diff --git a/gst-libs/gst/rtp/gstbasertppayload.c b/gst-libs/gst/rtp/gstbasertppayload.c
index f01b1a55f95681ed655afa1119819f499e4e7dc9..a5d07504e5e07a01143d927483f4e81439e12384 100644 (file)
--- a/gst-libs/gst/rtp/gstbasertppayload.c
+++ b/gst-libs/gst/rtp/gstbasertppayload.c
@@ -9,7 +9,7 @@
  * 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 
+ * Library General Public License for more
  */
 
 /**
@@ -54,7 +54,7 @@ enum
   LAST_SIGNAL
 };
 
-/* FIXME 0.11, a better default is the Ethernet MTU of 
+/* FIXME 0.11, a better default is the Ethernet MTU of
  * 1500 - sizeof(headers) as pointed out by marcelm in IRC:
  * So an Ethernet MTU of 1500, minus 60 for the max IP, minus 8 for UDP, gives
  * 1432 bytes or so.  And that should be adjusted downward further for other
@@ -418,7 +418,7 @@ no_function:
  * @payload: a #GstBaseRTPPayload
  * @media: the media type (typically "audio" or "video")
  * @dynamic: if the payload type is dynamic
- * @encoding_name: the encoding name 
+ * @encoding_name: the encoding name
  * @clock_rate: the clock rate of the media
  *
  * Set the rtp options of the payloader. These options will be set in the caps
@@ -621,6 +621,7 @@ typedef struct
   guint8 pt;
   GstCaps *caps;
   GstClockTime timestamp;
+  guint64 offset;
   guint32 rtptime;
 } HeaderData;
 
@@ -628,9 +629,10 @@ static GstBufferListItem
 find_timestamp (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
 {
   data->timestamp = GST_BUFFER_TIMESTAMP (*buffer);
+  data->offset = GST_BUFFER_OFFSET (*buffer);
 
-  /* stop when we find a timestamp */
-  if (data->timestamp != -1)
+  /* stop when we find a timestamp and duration */
+  if (data->timestamp != -1 && data->offset != -1)
     return GST_BUFFER_LIST_END;
   else
     return GST_BUFFER_LIST_CONTINUE;
@@ -644,6 +646,7 @@ set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
   gst_rtp_buffer_set_seq (*buffer, data->seqnum);
   gst_rtp_buffer_set_timestamp (*buffer, data->rtptime);
   gst_buffer_set_caps (*buffer, data->caps);
+  /* increment the seqnum for each buffer */
   data->seqnum++;
 
   return GST_BUFFER_LIST_SKIP_GROUP;
@@ -681,12 +684,17 @@ gst_basertppayload_prepare_push (GstBaseRTPPayload * payload,
         (GstBufferListFunc) find_timestamp, &data);
   } else {
     data.timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (obj));
+    data.offset = GST_BUFFER_OFFSET (GST_BUFFER_CAST (obj));
   }
 
   /* convert to RTP time */
-  if (GST_CLOCK_TIME_IS_VALID (data.timestamp)) {
+  if (data.offset != GST_BUFFER_OFFSET_NONE) {
+    /* if we have an offset, use that for making an RTP timestamp */
+    data.rtptime = payload->ts_base + data.offset;
+  } else if (GST_CLOCK_TIME_IS_VALID (data.timestamp)) {
     gint64 rtime;
 
+    /* no offset, use the gstreamer timestamp */
     rtime = gst_segment_to_running_time (&payload->segment, GST_FORMAT_TIME,
         data.timestamp);
 
@@ -764,7 +772,7 @@ gst_basertppayload_push_list (GstBaseRTPPayload * payload, GstBufferList * list)
  *
  * Push @buffer to the peer element of the payloader. The SSRC, payload type,
  * seqnum and timestamp of the RTP buffer will be updated first.
- * 
+ *
  * This function takes ownership of @buffer.
  *
  * Returns: a #GstFlowReturn.