baseaudiopay: fix timestamps on buffer lists

Fix the outgoing timestamps and RTP timestamps on outgoing buffers when using
buffer lists.

gst-libs/gst/rtp/gstbasertpaudiopayload.c

@@ -482,12 +482,11 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
 
 static GstFlowReturn
 gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
-    baseaudiopayload, GstBuffer * buffer)
+    baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
 {
   GstBaseRTPPayload *basepayload;
   GstBaseRTPAudioPayloadPrivate *priv;
   GstBuffer *outbuf;
-  GstClockTime timestamp;
   guint8 *payload;
   guint payload_len;
   GstFlowReturn ret;
@@ -496,7 +495,6 @@ gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
   basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
 
   payload_len = GST_BUFFER_SIZE (buffer);
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
 
   GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
       payload_len, GST_TIME_ARGS (timestamp));
@@ -607,7 +605,9 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
      * anything. */
     buffer = gst_adapter_take_buffer (adapter, payload_len);
 
-    ret = gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer);
+    ret =
+        gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer,
+        timestamp);
   } else {
     /* create buffer to hold the payload */
     outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
@@ -814,23 +814,22 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
   guint align;
   guint size;
   gboolean discont;
+  GstClockTime timestamp;
 
   ret = GST_FLOW_OK;
 
   payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
   priv = payload->priv;
 
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
   discont = GST_BUFFER_IS_DISCONT (buffer);
   if (discont) {
-    GstClockTime timestamp;
 
     GST_DEBUG_OBJECT (payload, "Got DISCONT");
     /* flush everything out of the adapter, mark DISCONT */
     ret = gst_base_rtp_audio_payload_flush (payload, -1, -1);
     priv->discont = TRUE;
 
-    timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
     /* get the distance between the timestamp gap and produce the same gap in
      * the RTP timestamps */
     if (priv->last_timestamp != -1 && timestamp != -1) {
@@ -878,7 +877,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
     /* If buffer fits on an RTP packet, let's just push it through
      * this will check against max_ptime and max_mtu */
     GST_DEBUG_OBJECT (payload, "Fast packet push");
-    ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer);
+    ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer, timestamp);
   } else {
     /* push the buffer in the adapter */
     gst_adapter_push (priv->adapter, buffer);