* 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
+#ifndef AVFORMAT_RTSP_H
+#define AVFORMAT_RTSP_H
#include <stdint.h>
#include "avformat.h"
#include "rtspcodes.h"
#include "rtpdec.h"
#include "network.h"
+#include "httpauth.h"
/**
* Network layer over which RTP/etc packet data will be transported.
RTSP_TRANSPORT_NB
};
+/**
+ * Transport mode for the RTSP data. This may be plain, or
+ * tunneled, which is done over HTTP.
+ */
+enum RTSPControlTransport {
+ RTSP_MODE_PLAIN, /**< Normal RTSP */
+ RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
+};
+
#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_NB_AUDIO_CHANNELS 1
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
#define RTSP_RTP_PORT_MIN 5000
#define RTSP_RTP_PORT_MAX 10000
* packets will be allowed to make before being discarded. */
int ttl;
- uint32_t destination; /**< destination IP address */
+ struct sockaddr_storage destination; /**< destination IP address */
+ char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
/** data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
* should be re-transmitted by the client in every RTSP command. */
char session_id[512];
+ /** the "Location:" field. This value is used to handle redirection.
+ */
+ char location[4096];
+
/** the "RealChallenge1:" field from the server */
char real_challenge[64];
* (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
* where platform is the output of $uname -msr | sed 's/ /-/g'. */
char server[64];
+
+ /** The "timeout" comes as part of the server response to the "SETUP"
+ * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
+ * time, in seconds, that the server will go without traffic over the
+ * RTSP/TCP connection before it closes the connection. To prevent
+ * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
+ * than this value. */
+ int timeout;
+
+ /** The "Notice" or "X-Notice" field value. See
+ * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
+ * for a complete list of supported values. */
+ int notice;
+
+ /** The "reason" is meant to specify better the meaning of the error code
+ * returned
+ */
+ char reason[256];
} RTSPMessageHeader;
/**
*/
enum RTSPClientState {
RTSP_STATE_IDLE, /**< not initialized */
- RTSP_STATE_PLAYING, /**< initialized and receiving data */
+ RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
+ RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
};
/**
/**
* Private data for the RTSP demuxer.
+ *
+ * @todo Use ByteIOContext instead of URLContext
*/
typedef struct RTSPState {
- URLContext *rtsp_hd; /* RTSP TCP connexion handle */
+ URLContext *rtsp_hd; /* RTSP TCP connection handle */
/** number of items in the 'rtsp_streams' variable */
int nb_rtsp_streams;
* identifier that the client should re-transmit in each RTSP command */
char session_id[512];
+ /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
+ * the server will go without traffic on the RTSP/TCP line before it
+ * closes the connection. */
+ int timeout;
+
+ /** timestamp of the last RTSP command that we sent to the RTSP server.
+ * This is used to calculate when to send dummy commands to keep the
+ * connection alive, in conjunction with timeout. */
+ int64_t last_cmd_time;
+
/** the negotiated data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
* of RTSPMessageHeader->real_challenge */
enum RTSPServerType server_type;
+ /** plaintext authorization line (username:password) */
+ char auth[128];
+
+ /** authentication state */
+ HTTPAuthState auth_state;
+
/** The last reply of the server to a RTSP command */
char last_reply[2048]; /* XXX: allocate ? */
/** stream setup during the last frame read. This is used to detect if
* we need to subscribe or unsubscribe to any new streams. */
- enum AVDiscard real_setup_cache[MAX_STREAMS];
+ enum AVDiscard *real_setup_cache;
+
+ /** current stream setup. This is a temporary buffer used to compare
+ * current setup to previous frame setup. */
+ enum AVDiscard *real_setup;
/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
* this is used to send the same "Unsubscribe:" if stream setup changed,
* before sending a new "Subscribe:" command. */
char last_subscription[1024];
//@}
+
+ /** The following are used for RTP/ASF streams */
+ //@{
+ /** ASF demuxer context for the embedded ASF stream from WMS servers */
+ AVFormatContext *asf_ctx;
+
+ /** cache for position of the asf demuxer, since we load a new
+ * data packet in the bytecontext for each incoming RTSP packet. */
+ uint64_t asf_pb_pos;
+ //@}
+
+ /** some MS RTSP streams contain a URL in the SDP that we need to use
+ * for all subsequent RTSP requests, rather than the input URI; in
+ * other cases, this is a copy of AVFormatContext->filename. */
+ char control_uri[1024];
+
+ /** Additional output handle, used when input and output are done
+ * separately, eg for HTTP tunneling. */
+ URLContext *rtsp_hd_out;
+
+ /** RTSP transport mode, such as plain or tunneled. */
+ enum RTSPControlTransport control_transport;
+
+ /* Number of RTCP BYE packets the RTSP session has received.
+ * An EOF is propagated back if nb_byes == nb_streams.
+ * This is reset after a seek. */
+ int nb_byes;
+
+ /** Reusable buffer for receiving packets */
+ uint8_t* recvbuf;
} RTSPState;
/**
*/
typedef struct RTSPStream {
URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
- void *transport_priv; /**< RTP/RDT parse context */
+ void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
/** corresponding stream index, if any. -1 if none (MPEG2TS case) */
int stream_index;
/** The following are used only in SDP, not RTSP */
//@{
int sdp_port; /**< port (from SDP content) */
- struct in_addr sdp_ip; /**< IP address (from SDP content) */
+ struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
int sdp_payload_type; /**< payload type */
//@}
- /** rtp payload parsing infos from SDP (i.e. mapping between private
- * payload IDs and media-types (string), so that we can derive what
- * type of payload we're dealing with (and how to parse it). */
- RTPPayloadData rtp_payload_data;
-
/** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
//@{
/** handler structure */
//@}
} RTSPStream;
-int rtsp_init(void);
-void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf);
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
+ HTTPAuthState *auth_state);
-#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);
+/**
+ * Send a command to the RTSP server without waiting for the reply.
+ *
+ * @param s RTSP (de)muxer context
+ * @param method the method for the request
+ * @param url the target url for the request
+ * @param headers extra header lines to include in the request
+ * @param send_content if non-null, the data to send as request body content
+ * @param send_content_length the length of the send_content data, or 0 if
+ * send_content is null
+ *
+ * @return zero if success, nonzero otherwise
+ */
+int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *headers,
+ const unsigned char *send_content,
+ int send_content_length);
+/**
+ * Send a command to the RTSP server without waiting for the reply.
+ *
+ * @see rtsp_send_cmd_with_content_async
+ */
+int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
+ const char *url, const char *headers);
+
+/**
+ * Send a command to the RTSP server and wait for the reply.
+ *
+ * @param s RTSP (de)muxer context
+ * @param method the method for the request
+ * @param url the target url for the request
+ * @param headers extra header lines to include in the request
+ * @param reply pointer where the RTSP message header will be stored
+ * @param content_ptr pointer where the RTSP message body, if any, will
+ * be stored (length is in reply)
+ * @param send_content if non-null, the data to send as request body content
+ * @param send_content_length the length of the send_content data, or 0 if
+ * send_content is null
+ *
+ * @return zero if success, nonzero otherwise
+ */
+int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
+ const char *method, const char *url,
+ const char *headers,
+ RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ const unsigned char *send_content,
+ int send_content_length);
+
+/**
+ * Send a command to the RTSP server and wait for the reply.
+ *
+ * @see rtsp_send_cmd_with_content
+ */
+int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
+ const char *url, const char *headers,
+ RTSPMessageHeader *reply, unsigned char **content_ptr);
+
+/**
+ * Read a RTSP message from the server, or prepare to read data
+ * packets if we're reading data interleaved over the TCP/RTSP
+ * connection as well.
+ *
+ * @param s RTSP (de)muxer context
+ * @param reply pointer where the RTSP message header will be stored
+ * @param content_ptr pointer where the RTSP message body, if any, will
+ * be stored (length is in reply)
+ * @param return_on_interleaved_data whether the function may return if we
+ * encounter a data marker ('$'), which precedes data
+ * packets over interleaved TCP/RTSP connections. If this
+ * is set, this function will return 1 after encountering
+ * a '$'. If it is not set, the function will skip any
+ * data packets (if they are encountered), until a reply
+ * has been fully parsed. If no more data is available
+ * without parsing a reply, it will return an error.
+ *
+ * @return 1 if a data packets is ready to be received, -1 on error,
+ * and 0 on success.
+ */
+int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ int return_on_interleaved_data);
+
+/**
+ * Skip a RTP/TCP interleaved packet.
+ */
+void ff_rtsp_skip_packet(AVFormatContext *s);
+
+/**
+ * Connect to the RTSP server and set up the individual media streams.
+ * This can be used for both muxers and demuxers.
+ *
+ * @param s RTSP (de)muxer context
+ *
+ * @return 0 on success, < 0 on error. Cleans up all allocations done
+ * within the function on error.
+ */
+int ff_rtsp_connect(AVFormatContext *s);
+
+/**
+ * Close and free all streams within the RTSP (de)muxer
+ *
+ * @param s RTSP (de)muxer context
+ */
+void ff_rtsp_close_streams(AVFormatContext *s);
+
+/**
+ * Close all connection handles within the RTSP (de)muxer
+ *
+ * @param rt RTSP (de)muxer context
+ */
+void ff_rtsp_close_connections(AVFormatContext *rt);
-#endif /* FFMPEG_RTSP_H */
+#endif /* AVFORMAT_RTSP_H */
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