rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpgsmpay.c

diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
index 3765805..25fa0fa 100644 (file)
--- a/gst/rtp/gstrtpgsmpay.c
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -74,10 +74,10 @@ gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
   gstelement_class = (GstElementClass *) klass;
   gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
 
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_gsm_pay_sink_template);
+  gst_element_class_add_static_pad_template (gstelement_class,
+      &gst_rtp_gsm_pay_src_template);
 
   gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
       "Codec/Payloader/Network/RTP",
@@ -145,12 +145,14 @@ gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
   if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
     goto too_big;
 
-  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
   /* copy timestamp and duration */
   GST_BUFFER_PTS (outbuf) = timestamp;
   GST_BUFFER_DURATION (outbuf) = duration;
 
+  gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
+
   /* append payload */
   outbuf = gst_buffer_append (outbuf, buffer);