rtp: Update codes based on 1.18.4

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

diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c
index 8dcd1f1..6f086b0 100644 (file)
--- a/gst/rtp/gstrtpspeexpay.c
+++ b/gst/rtp/gstrtpspeexpay.c
@@ -279,8 +279,8 @@ gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
   duration = GST_BUFFER_DURATION (buffer);
 
   /* FIXME, only one SPEEX frame per RTP packet for now */
+  outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
 
-  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
   /* FIXME, assert for now */
   g_assert (gst_buffer_get_size (buffer) <=
       GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));
@@ -289,8 +289,7 @@ gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
   GST_BUFFER_PTS (outbuf) = timestamp;
   GST_BUFFER_DURATION (outbuf) = duration;
 
-  gst_rtp_copy_meta (GST_ELEMENT_CAST (basepayload), outbuf, buffer,
-      g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
+  gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
   outbuf = gst_buffer_append (outbuf, buffer);
   buffer = NULL;