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

1 files changed, 60 insertions, 0 deletions

diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 611f5c3aae..36297e98c0 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -24,6 +24,8 @@
 #include <stdint.h>
 #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,