Update public FFmpeg header files from r16647 to r18286.

third_party/ffmpeg/include/libavformat/rtsp.h

-/*
- * RTSP definitions
- * Copyright (c) 2002 Fabrice Bellard.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-#ifndef FFMPEG_RTSP_H
-#define FFMPEG_RTSP_H
-
-#include <stdint.h>
-#include "avformat.h"
-#include "rtspcodes.h"
-#include "rtp.h"
-#include "network.h"
-
-enum RTSPLowerTransport {
-    RTSP_LOWER_TRANSPORT_UDP = 0,
-    RTSP_LOWER_TRANSPORT_TCP = 1,
-    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2,
-    /**
-     * This is not part of public API and shouldn't be used outside of ffmpeg.
-     */
-    RTSP_LOWER_TRANSPORT_LAST
-};
-
-#define RTSP_DEFAULT_PORT   554
-#define RTSP_MAX_TRANSPORTS 8
-#define RTSP_TCP_MAX_PACKET_SIZE 1472
-#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
-#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
-#define RTSP_RTP_PORT_MIN 5000
-#define RTSP_RTP_PORT_MAX 10000
-
-typedef struct RTSPTransportField {
-    int interleaved_min, interleaved_max;  /**< interleave ids, if TCP transport */
-    int port_min, port_max; /**< RTP ports */
-    int client_port_min, client_port_max; /**< RTP ports */
-    int server_port_min, server_port_max; /**< RTP ports */
-    int ttl; /**< ttl value */
-    uint32_t destination; /**< destination IP address */
-    int transport;
-    enum RTSPLowerTransport lower_transport;
-} RTSPTransportField;
-
-typedef struct RTSPHeader {
-    int content_length;
-    enum RTSPStatusCode status_code; /**< response code from server */
-    int nb_transports;
-    /** in AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
-    int64_t range_start, range_end;
-    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
-    int seq; /**< sequence number */
-    char session_id[512];
-    char real_challenge[64]; /**< the RealChallenge1 field from the server */
-    char server[64];
-} RTSPHeader;
-
-enum RTSPClientState {
-    RTSP_STATE_IDLE,
-    RTSP_STATE_PLAYING,
-    RTSP_STATE_PAUSED,
-};
-
-enum RTSPServerType {
-    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
-    RTSP_SERVER_REAL, /**< Realmedia-style server */
-    RTSP_SERVER_WMS,  /**< Windows Media 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,
-    RTSP_ACTION_SERVER_TEARDOWN,
-    RTSP_ACTION_CLIENT_SETUP,
-    RTSP_ACTION_CLIENT_TEARDOWN,
-};
-
-typedef struct RTSPActionServerSetup {
-    uint32_t ipaddr;
-    char transport_option[512];
-} RTSPActionServerSetup;
-
-typedef int FFRTSPCallback(enum RTSPCallbackAction action,
-                           const char *session_id,
-                           char *buf, int buf_size,
-                           void *arg);
-
-int rtsp_init(void);
-void rtsp_parse_line(RTSPHeader *reply, const char *buf);
-
-#if LIBAVFORMAT_VERSION_INT < (53 << 16)
-extern int rtsp_default_protocols;
-#endif
-extern int rtsp_rtp_port_min;
-extern int rtsp_rtp_port_max;
-
-int rtsp_pause(AVFormatContext *s);
-int rtsp_resume(AVFormatContext *s);
-
-#endif /* FFMPEG_RTSP_H */