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author | Ronald S. Bultje <rsbultje@gmail.com> | 2009-01-07 14:53:04 +0000 |
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committer | Ronald S. Bultje <rsbultje@gmail.com> | 2009-01-07 14:53:04 +0000 |
commit | 74272b1c0cb4b90da324eec78c7dbe74e3993a77 (patch) | |
tree | b81b0e87e09d3609e96042e68df5cdb1bf412bca /libavformat/rtsp.c | |
parent | b0d75d6036d8762a0019a9a976e913b9ac9bf7f8 (diff) | |
download | ffmpeg-74272b1c0cb4b90da324eec78c7dbe74e3993a77.tar.gz |
Export RTSPState and RTSPStream from rtsp.c into rtsp.h. This allows future
access to these structures in functions that will be located in rtp_asf.c.
See "[PATCH] RTSP-MS 2/15: export RTSPState and RTSPStream" mailinglist
thread.
Originally committed as revision 16471 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtsp.c')
-rw-r--r-- | libavformat/rtsp.c | 58 |
1 files changed, 0 insertions, 58 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index f0c6e1b3ab..2ba1755b28 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -39,64 +39,6 @@ //#define DEBUG //#define DEBUG_RTP_TCP -enum RTSPClientState { - RTSP_STATE_IDLE, - RTSP_STATE_PLAYING, - RTSP_STATE_PAUSED, -}; - -enum RTSPServerType { - RTSP_SERVER_RTP, /*< Standard-compliant RTP-server */ - RTSP_SERVER_REAL, /*< Realmedia-style server */ - RTSP_SERVER_LAST -}; - -enum RTSPTransport { - RTSP_TRANSPORT_RTP, - RTSP_TRANSPORT_RDT, - RTSP_TRANSPORT_LAST -}; - -typedef struct RTSPState { - URLContext *rtsp_hd; /* RTSP TCP connexion handle */ - int nb_rtsp_streams; - struct RTSPStream **rtsp_streams; - - enum RTSPClientState state; - int64_t seek_timestamp; - - /* XXX: currently we use unbuffered input */ - // ByteIOContext rtsp_gb; - int seq; /* RTSP command sequence number */ - char session_id[512]; - enum RTSPTransport transport; - enum RTSPLowerTransport lower_transport; - enum RTSPServerType server_type; - char last_reply[2048]; /* XXX: allocate ? */ - void *cur_tx; - int need_subscription; - enum AVDiscard real_setup_cache[MAX_STREAMS]; - char last_subscription[1024]; -} RTSPState; - -typedef struct RTSPStream { - URLContext *rtp_handle; /* RTP stream handle */ - void *tx_ctx; /* RTP/RDT parse context */ - - int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ - int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ - char control_url[1024]; /* url for this stream (from SDP) */ - - int sdp_port; /* port (from SDP content - not used in RTSP) */ - struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ - int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ - int sdp_payload_type; /* payload type - only used in SDP */ - RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */ - - RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) - PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) -} RTSPStream; - static int rtsp_read_play(AVFormatContext *s); /* XXX: currently, the only way to change the protocols consists in |