From 74272b1c0cb4b90da324eec78c7dbe74e3993a77 Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Wed, 7 Jan 2009 14:53:04 +0000 Subject: [PATCH] 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 --- libavformat/rtsp.c | 58 ---------------------------------------------------- libavformat/rtsp.h | 60 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 60 insertions(+), 58 deletions(-) diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index f0c6e1b3a..2ba1755b2 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 diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 611f5c3aa..36297e98c 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -24,6 +24,8 @@ #include #include "avformat.h" #include "rtspcodes.h" +#include "rtp.h" +#include "network.h" enum RTSPLowerTransport { RTSP_LOWER_TRANSPORT_UDP = 0, @@ -66,6 +68,64 @@ typedef struct RTSPHeader { char real_challenge[64]; /**< the RealChallenge1 field from the server */ } RTSPHeader; +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; + /** the callback can be used to extend the connection setup/teardown step */ enum RTSPCallbackAction { RTSP_ACTION_SERVER_SETUP, -- 2.11.0