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;
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);
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;