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authorRonald S. Bultje <rsbultje@gmail.com>2009-01-07 14:53:04 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2009-01-07 14:53:04 +0000
commit74272b1c0cb4b90da324eec78c7dbe74e3993a77 (patch)
treeb81b0e87e09d3609e96042e68df5cdb1bf412bca /libavformat/rtsp.c
parentb0d75d6036d8762a0019a9a976e913b9ac9bf7f8 (diff)
downloadffmpeg-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.c58
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