Rename RTSP_SERVER_RDT to RTSP_SERVER_REAL, because RDT (the transport

[ffmpeg] / libavformat / rtp.h

diff --git a/libavformat/rtp.h b/libavformat/rtp.h
index 4f6251bf34404790d68232713cadc6f622be8253..0eb95653cbacdbd3a4af33abd1173ad30202996d 100644 (file)
--- a/libavformat/rtp.h
+++ b/libavformat/rtp.h
@@ -18,10 +18,10 @@
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
-#ifndef FFMPEG_RTP_H
-#define FFMPEG_RTP_H
+#ifndef AVFORMAT_RTP_H
+#define AVFORMAT_RTP_H
 
-#include "avcodec.h"
+#include "libavcodec/avcodec.h"
 #include "avformat.h"
 
 #define RTP_MIN_PACKET_LENGTH 12
@@ -85,16 +85,6 @@ typedef struct rtp_payload_data_s
     int cur_au_index;
 } rtp_payload_data_t;
 
-typedef struct AVRtpPayloadType_s
-{
-    int pt;
-    const char enc_name[50]; /* XXX: why 50 ? */
-    enum CodecType codec_type;
-    enum CodecID codec_id;
-    int clock_rate;
-    int audio_channels;
-} AVRtpPayloadType_t;
-
 #if 0
 typedef enum {
   RTCP_SR   = 200,
@@ -120,5 +110,4 @@ typedef enum {
 } rtcp_sdes_type_t;
 #endif
 
-extern AVRtpPayloadType_t AVRtpPayloadTypes[];
-#endif /* FFMPEG_RTP_H */
+#endif /* AVFORMAT_RTP_H */