1 files changed, 0 insertions, 127 deletions
diff --git a/contrib/ffmpeg/libavformat/rtp.h b/contrib/ffmpeg/libavformat/rtp.h
deleted file mode 100644
index fec763051..000000000
--- a/contrib/ffmpeg/libavformat/rtp.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * RTP definitions
- * Copyright (c) 2002 Fabrice Bellard.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-#ifndef RTP_H
-#define RTP_H
-
-#define RTP_MIN_PACKET_LENGTH 12
-#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
-
-int rtp_init(void);
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
-
-/** return < 0 if unknown payload type */
-int rtp_get_payload_type(AVCodecContext *codec);
-
-typedef struct RTPDemuxContext RTPDemuxContext;
-typedef struct rtp_payload_data_s rtp_payload_data_s;
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     const uint8_t *buf, int len);
-void rtp_parse_close(RTPDemuxContext *s);
-
-extern AVOutputFormat rtp_muxer;
-extern AVInputFormat rtp_demuxer;
-
-int rtp_get_local_port(URLContext *h);
-int rtp_set_remote_url(URLContext *h, const char *uri);
-void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
-
-/**
- * some rtp servers assume client is dead if they don't hear from them...
- * so we send a Receiver Report to the provided ByteIO context
- * (we don't have access to the rtcp handle from here)
- */
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
-
-extern URLProtocol rtp_protocol;
-
-#define RTP_PT_PRIVATE 96
-#define RTP_VERSION 2
-#define RTP_MAX_SDES 256   /**< maximum text length for SDES */
-
-/* RTCP paquets use 0.5 % of the bandwidth */
-#define RTCP_TX_RATIO_NUM 5
-#define RTCP_TX_RATIO_DEN 1000
-
-/** Structure listing useful vars to parse RTP packet payload*/
-typedef struct rtp_payload_data_s
-{
-    int sizelength;
-    int indexlength;
-    int indexdeltalength;
-    int profile_level_id;
-    int streamtype;
-    int objecttype;
-    char *mode;
-
-    /** mpeg 4 AU headers */
-    struct AUHeaders {
-        int size;
-        int index;
-        int cts_flag;
-        int cts;
-        int dts_flag;
-        int dts;
-        int rap_flag;
-        int streamstate;
-    } *au_headers;
-    int nb_au_headers;
-    int au_headers_length_bytes;
-    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,
-  RTCP_RR   = 201,
-  RTCP_SDES = 202,
-  RTCP_BYE  = 203,
-  RTCP_APP  = 204
-} rtcp_type_t;
-
-typedef enum {
-  RTCP_SDES_END    =  0,
-  RTCP_SDES_CNAME  =  1,
-  RTCP_SDES_NAME   =  2,
-  RTCP_SDES_EMAIL  =  3,
-  RTCP_SDES_PHONE  =  4,
-  RTCP_SDES_LOC    =  5,
-  RTCP_SDES_TOOL   =  6,
-  RTCP_SDES_NOTE   =  7,
-  RTCP_SDES_PRIV   =  8,
-  RTCP_SDES_IMG    =  9,
-  RTCP_SDES_DOOR   = 10,
-  RTCP_SDES_SOURCE = 11
-} rtcp_sdes_type_t;
-#endif
-
-extern AVRtpPayloadType_t AVRtpPayloadTypes[];
-#endif /* RTP_H */