lavf: rename avf_sdp_create to av_sdp_create.

[ffmpeg] / libavformat / rtpdec.c

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index bbd5f338ed441ad8885700fd0f42d70e02a019be..43305a30c794543a1b59586f1252dd6c877a84b9 100644 (file)
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -25,6 +25,7 @@
 #include "libavcodec/get_bits.h"
 #include "avformat.h"
 #include "mpegts.h"
+#include "url.h"
 
 #include <unistd.h>
 #include <strings.h>
@@ -41,7 +42,7 @@
          buffer to 'rtp_write_packet' contains all the packets for ONE
          frame. Each packet should have a four byte header containing
          the length in big endian format (same trick as
-         'url_open_dyn_packet_buf')
+         'ffio_open_dyn_packet_buf')
 */
 
 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
@@ -325,8 +326,8 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
     if ((len > 0) && buf) {
         int result;
         av_dlog(s->ic, "sending %d bytes of RR\n", len);
-        result= url_write(s->rtp_ctx, buf, len);
-        av_dlog(s->ic, "result from url_write: %d\n", result);
+        result= ffurl_write(s->rtp_ctx, buf, len);
+        av_dlog(s->ic, "result from ffurl_write: %d\n", result);
         av_free(buf);
     }
     return 0;
@@ -351,7 +352,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
     avio_flush(pb);
     len = avio_close_dyn_buf(pb, &buf);
     if ((len > 0) && buf)
-        url_write(rtp_handle, buf, len);
+        ffurl_write(rtp_handle, buf, len);
     av_free(buf);
 
     /* Send a minimal RTCP RR */
@@ -366,7 +367,7 @@ void rtp_send_punch_packets(URLContext* rtp_handle)
     avio_flush(pb);
     len = avio_close_dyn_buf(pb, &buf);
     if ((len > 0) && buf)
-        url_write(rtp_handle, buf, len);
+        ffurl_write(rtp_handle, buf, len);
     av_free(buf);
 }