mirror of https://git.ffmpeg.org/ffmpeg.git
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
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@ -39,64 +39,6 @@
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//#define DEBUG
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//#define DEBUG
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//#define DEBUG_RTP_TCP
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//#define DEBUG_RTP_TCP
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enum RTSPClientState {
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RTSP_STATE_IDLE,
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RTSP_STATE_PLAYING,
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RTSP_STATE_PAUSED,
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};
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enum RTSPServerType {
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RTSP_SERVER_RTP, /*< Standard-compliant RTP-server */
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RTSP_SERVER_REAL, /*< Realmedia-style server */
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RTSP_SERVER_LAST
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};
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enum RTSPTransport {
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RTSP_TRANSPORT_RTP,
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RTSP_TRANSPORT_RDT,
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RTSP_TRANSPORT_LAST
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};
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typedef struct RTSPState {
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams;
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enum RTSPClientState state;
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int64_t seek_timestamp;
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/* XXX: currently we use unbuffered input */
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// ByteIOContext rtsp_gb;
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int seq; /* RTSP command sequence number */
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char session_id[512];
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enum RTSPTransport transport;
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enum RTSPLowerTransport lower_transport;
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enum RTSPServerType server_type;
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char last_reply[2048]; /* XXX: allocate ? */
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void *cur_tx;
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int need_subscription;
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enum AVDiscard real_setup_cache[MAX_STREAMS];
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char last_subscription[1024];
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} RTSPState;
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typedef struct RTSPStream {
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URLContext *rtp_handle; /* RTP stream handle */
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void *tx_ctx; /* RTP/RDT parse context */
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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char control_url[1024]; /* url for this stream (from SDP) */
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int sdp_port; /* port (from SDP content - not used in RTSP) */
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
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int sdp_payload_type; /* payload type - only used in SDP */
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RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
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PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
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} RTSPStream;
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static int rtsp_read_play(AVFormatContext *s);
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static int rtsp_read_play(AVFormatContext *s);
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/* XXX: currently, the only way to change the protocols consists in
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/* XXX: currently, the only way to change the protocols consists in
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@ -24,6 +24,8 @@
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#include <stdint.h>
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#include <stdint.h>
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#include "avformat.h"
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#include "avformat.h"
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#include "rtspcodes.h"
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#include "rtspcodes.h"
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#include "rtp.h"
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#include "network.h"
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enum RTSPLowerTransport {
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enum RTSPLowerTransport {
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RTSP_LOWER_TRANSPORT_UDP = 0,
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RTSP_LOWER_TRANSPORT_UDP = 0,
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@ -66,6 +68,64 @@ typedef struct RTSPHeader {
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char real_challenge[64]; /**< the RealChallenge1 field from the server */
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char real_challenge[64]; /**< the RealChallenge1 field from the server */
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} RTSPHeader;
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} RTSPHeader;
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enum RTSPClientState {
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RTSP_STATE_IDLE,
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RTSP_STATE_PLAYING,
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RTSP_STATE_PAUSED,
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};
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enum RTSPServerType {
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RTSP_SERVER_RTP, /*< Standard-compliant RTP-server */
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RTSP_SERVER_REAL, /*< Realmedia-style server */
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RTSP_SERVER_LAST
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};
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enum RTSPTransport {
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RTSP_TRANSPORT_RTP,
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RTSP_TRANSPORT_RDT,
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RTSP_TRANSPORT_LAST
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};
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typedef struct RTSPState {
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URLContext *rtsp_hd; /* RTSP TCP connexion handle */
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int nb_rtsp_streams;
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struct RTSPStream **rtsp_streams;
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enum RTSPClientState state;
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int64_t seek_timestamp;
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/* XXX: currently we use unbuffered input */
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// ByteIOContext rtsp_gb;
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int seq; /* RTSP command sequence number */
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char session_id[512];
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enum RTSPTransport transport;
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enum RTSPLowerTransport lower_transport;
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enum RTSPServerType server_type;
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char last_reply[2048]; /* XXX: allocate ? */
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void *cur_tx;
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int need_subscription;
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enum AVDiscard real_setup_cache[MAX_STREAMS];
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char last_subscription[1024];
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} RTSPState;
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typedef struct RTSPStream {
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URLContext *rtp_handle; /* RTP stream handle */
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void *tx_ctx; /* RTP/RDT parse context */
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int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
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int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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char control_url[1024]; /* url for this stream (from SDP) */
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int sdp_port; /* port (from SDP content - not used in RTSP) */
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struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
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int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
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int sdp_payload_type; /* payload type - only used in SDP */
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RTPPayloadData rtp_payload_data; /* rtp payload parsing infos from SDP */
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RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
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PayloadContext *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
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} RTSPStream;
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/** the callback can be used to extend the connection setup/teardown step */
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/** the callback can be used to extend the connection setup/teardown step */
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enum RTSPCallbackAction {
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enum RTSPCallbackAction {
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RTSP_ACTION_SERVER_SETUP,
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RTSP_ACTION_SERVER_SETUP,
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