rtp: Update codes based on 1.18.4

[platform/upstream/gst-plugins-good.git] / gst / rtp / gstrtpmp4apay.c

diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c
index 64b4baa..ab1eeb5 100644 (file)
--- a/gst/rtp/gstrtpmp4apay.c
+++ b/gst/rtp/gstrtpmp4apay.c
@@ -67,9 +67,10 @@ static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
     payload, GstBuffer * buffer);
 
 #define gst_rtp_mp4a_pay_parent_class parent_class
-G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD)
+G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD);
 
-     static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
+static void
+gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
@@ -407,7 +408,8 @@ gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
         packet_len, payload_len);
 
     /* create buffer to hold the payload. */
-    outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
+    outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
+        header_len, 0, 0);
 
     /* copy payload */
     gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
@@ -435,14 +437,13 @@ gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
         offset, payload_len);
 
     /* join memory parts */
-    gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4apay), outbuf, paybuf,
-        g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
+    gst_rtp_copy_audio_meta (rtpmp4apay, outbuf, paybuf);
     outbuf = gst_buffer_append (outbuf, paybuf);
     gst_buffer_list_add (list, outbuf);
     offset += payload_len;
     size -= payload_len;
 
-    /* copy incomming timestamp (if any) to outgoing buffers */
+    /* copy incoming timestamp (if any) to outgoing buffers */
     GST_BUFFER_PTS (outbuf) = timestamp;
 
     fragmented = TRUE;