[MPlayer-cvslog] r18833 - in trunk: cfg-common.h libmpdemux/Makefile libmpdemux/librtsp/rtsp_rtp.c libmpdemux/librtsp/rtsp_rtp.h libmpdemux/librtsp/rtsp_session.c libmpdemux/librtsp/rtsp_session.h libmpdemux/rtp.c libmpdemux/rtp.h
ben
subversion at mplayerhq.hu
Mon Jun 26 23:27:58 CEST 2006
Author: ben
Date: Mon Jun 26 23:27:57 2006
New Revision: 18833
Added:
trunk/libmpdemux/librtsp/rtsp_rtp.c
trunk/libmpdemux/librtsp/rtsp_rtp.h
Modified:
trunk/cfg-common.h
trunk/libmpdemux/Makefile
trunk/libmpdemux/librtsp/rtsp_session.c
trunk/libmpdemux/librtsp/rtsp_session.h
trunk/libmpdemux/rtp.c
trunk/libmpdemux/rtp.h
Log:
added new native rtsp demuxer code for mpeg-ts over rtp (now both real and non-real servers should be handled)
Modified: trunk/cfg-common.h
==============================================================================
--- trunk/cfg-common.h (original)
+++ trunk/cfg-common.h Mon Jun 26 23:27:57 2006
@@ -70,12 +70,11 @@
{"sdp", "-sdp is obsolete, use sdp://file instead.\n", CONF_TYPE_PRINT, 0, 0, 0, NULL},
// -rtsp-stream-over-tcp option, specifying TCP streaming of RTP/RTCP
{"rtsp-stream-over-tcp", &rtspStreamOverTCP, CONF_TYPE_FLAG, 0, 0, 1, NULL},
- {"rtsp-port", &rtsp_port, CONF_TYPE_INT, CONF_RANGE, -1, 65535, NULL},
#else
{"rtsp-stream-over-tcp", "RTSP support requires the \"LIVE555 Streaming Media\" libraries.\n", CONF_TYPE_PRINT, CONF_NOCFG, 0, 0, NULL},
- {"rtsp-port", "RTSP support requires the \"LIVE555 Streaming Media\" libraries.\n", CONF_TYPE_PRINT, CONF_NOCFG, 0, 0, NULL},
#endif
-
+ {"rtsp-port", &rtsp_port, CONF_TYPE_INT, CONF_RANGE, -1, 65535, NULL},
+
// ------------------------- demuxer options --------------------
// number of frames to play/convert
@@ -415,8 +414,8 @@
#ifdef STREAMING_LIVE555
extern int rtspStreamOverTCP;
-extern int rtsp_port;
#endif
+extern int rtsp_port;
extern int audio_stream_cache;
Modified: trunk/libmpdemux/Makefile
==============================================================================
--- trunk/libmpdemux/Makefile (original)
+++ trunk/libmpdemux/Makefile Mon Jun 26 23:27:57 2006
@@ -139,6 +139,7 @@
realrtsp/xbuffer.c \
SRCS += librtsp/rtsp.c \
+ librtsp/rtsp_rtp.c \
librtsp/rtsp_session.c \
SRCS += freesdp/common.c \
Added: trunk/libmpdemux/librtsp/rtsp_rtp.c
==============================================================================
--- (empty file)
+++ trunk/libmpdemux/librtsp/rtsp_rtp.c Mon Jun 26 23:27:57 2006
@@ -0,0 +1,685 @@
+/*
+ * Copyright (C) 2006 Benjamin Zores
+ * based on the Freebox patch for xine by Vincent Mussard
+ * but with many enhancements for better RTSP RFC compliance.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <pthread.h>
+#include <netdb.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <string.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <sys/types.h>
+
+#ifndef HAVE_WINSOCK2
+#include <netinet/in.h>
+#include <sys/socket.h>
+#include <arpa/inet.h>
+#define closesocket close
+#else
+#include <winsock2.h>
+#include <ws2tcpip.h>
+#endif
+
+#include "mp_msg.h"
+#include "rtsp.h"
+#include "rtsp_rtp.h"
+#include "rtsp_session.h"
+#include "../freesdp/common.h"
+#include "../freesdp/parser.h"
+
+#define RTSP_DEFAULT_PORT 31336
+#define MAX_LENGTH 256
+
+#define RTSP_ACCEPT_SDP "Accept: application/sdp"
+#define RTSP_CONTENT_LENGTH "Content-length"
+#define RTSP_CONTENT_TYPE "Content-Type"
+#define RTSP_APPLICATION_SDP "application/sdp"
+#define RTSP_RANGE "Range: "
+#define RTSP_NPT_NOW "npt=now-"
+#define RTSP_MEDIA_CONTAINER_MPEG_TS "33"
+#define RTSP_TRANSPORT_REQUEST "Transport: RTP/AVP;%s;%s%i-%i;mode=\"PLAY\""
+
+#define RTSP_TRANSPORT_MULTICAST "multicast"
+#define RTSP_TRANSPORT_UNICAST "unicast"
+
+#define RTSP_MULTICAST_PORT "port="
+#define RTSP_UNICAST_CLIENT_PORT "client_port="
+#define RTSP_UNICAST_SERVER_PORT "server_port="
+#define RTSP_SETUP_DESTINATION "destination="
+
+#define RTSP_SESSION "Session"
+#define RTSP_TRANSPORT "Transport"
+
+/* hardcoded RTCP RR - this is _NOT_ RFC compliant */
+#define RTCP_RR_SIZE 32
+#define RTCP_RR "\201\311\0\7(.JD\31+\306\343\0\0\0\0\0\0/E\0\0\2&\0\0\0\0\0\0\0\0\201"
+#define RTCP_SEND_FREQUENCY 1024
+
+int rtsp_port = 0;
+
+void
+rtcp_send_rr (rtsp_t *s, struct rtp_rtsp_session_t *st)
+{
+ if (st->rtcp_socket == -1)
+ return;
+
+ /* send RTCP RR every RTCP_SEND_FREQUENCY packets
+ * FIXME : NOT CORRECT, HARDCODED, BUT MAKES SOME SERVERS HAPPY
+ * not rfc compliant
+ * http://www.faqs.org/rfcs/rfc1889.html chapter 6 for RTCP
+ */
+
+ if (st->count == RTCP_SEND_FREQUENCY)
+ {
+ char rtcp_content[RTCP_RR_SIZE];
+ strcpy (rtcp_content, RTCP_RR);
+ send (st->rtcp_socket, rtcp_content, RTCP_RR_SIZE, 0);
+
+ /* ping RTSP server to keep connection alive.
+ we use OPTIONS instead of PING as not all servers support it */
+ rtsp_request_options (s, "*");
+ st->count = 0;
+ }
+ else
+ st->count++;
+}
+
+static struct rtp_rtsp_session_t *
+rtp_session_new (void)
+{
+ struct rtp_rtsp_session_t *st = NULL;
+
+ st = malloc (sizeof (struct rtp_rtsp_session_t));
+
+ st->rtp_socket = -1;
+ st->rtcp_socket = -1;
+ st->control_url = NULL;
+ st->count = 0;
+
+ return st;
+}
+
+void
+rtp_session_free (struct rtp_rtsp_session_t *st)
+{
+ if (!st)
+ return;
+
+ if (st->rtp_socket != -1)
+ close (st->rtp_socket);
+ if (st->rtcp_socket != -1)
+ close (st->rtcp_socket);
+
+ if (st->control_url)
+ free (st->control_url);
+ free (st);
+}
+
+static void
+rtp_session_set_fd (struct rtp_rtsp_session_t *st,
+ int rtp_sock, int rtcp_sock)
+{
+ if (!st)
+ return;
+
+ st->rtp_socket = rtp_sock;
+ st->rtcp_socket = rtcp_sock;
+}
+
+static int
+parse_port (const char *line, const char *param,
+ int *rtp_port, int *rtcp_port)
+{
+ char *parse1;
+ char *parse2;
+ char *parse3;
+
+ char *line_copy = strdup (line);
+
+ parse1 = strstr (line_copy, param);
+
+ if (parse1)
+ {
+ parse2 = strstr (parse1, "-");
+
+ if (parse2)
+ {
+ parse3 = strstr (parse2, ";");
+
+ if (parse3)
+ parse3[0] = 0;
+
+ parse2[0] = 0;
+ }
+ else
+ {
+ free (line_copy);
+ return 0;
+ }
+ }
+ else
+ {
+ free (line_copy);
+ return 0;
+ }
+
+ *rtp_port = atoi (parse1 + strlen (param));
+ *rtcp_port = atoi (parse2 + 1);
+
+ free (line_copy);
+
+ return 1;
+}
+
+static char *
+parse_destination (const char *line)
+{
+ char *parse1;
+ char *parse2;
+
+ char *dest = NULL;
+ char *line_copy = strdup (line);
+ int len;
+
+ parse1 = strstr (line_copy, RTSP_SETUP_DESTINATION);
+ if (!parse1)
+ {
+ free (line_copy);
+ return NULL;
+ }
+
+ parse2 = strstr (parse1, ";");
+ if (!parse2)
+ {
+ free (line_copy);
+ return NULL;
+ }
+
+ len = strlen (parse1) - strlen (parse2)
+ - strlen (RTSP_SETUP_DESTINATION) + 1;
+ dest = (char *) malloc (len + 1);
+ snprintf (dest, len, parse1 + strlen (RTSP_SETUP_DESTINATION));
+ free (line_copy);
+
+ return dest;
+}
+
+static int
+rtcp_connect (int client_port, int server_port, const char* server_hostname)
+{
+ struct sockaddr_in sin;
+ struct hostent *hp;
+ int s;
+
+ if (client_port <= 1023)
+ return -1;
+
+ s = socket (PF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (s == -1)
+ return -1;
+
+ hp = gethostbyname (server_hostname);
+ if (!hp)
+ {
+ close (s);
+ return -1;
+ }
+
+ sin.sin_family = AF_INET;
+ sin.sin_addr.s_addr = INADDR_ANY;
+ sin.sin_port = htons (client_port);
+
+ if (bind (s, (struct sockaddr *) &sin, sizeof (sin)))
+ {
+#ifndef HAVE_WINSOCK2
+ if (errno != EINPROGRESS)
+#else
+ if (WSAGetLastError() != WSAEINPROGRESS)
+#endif
+ {
+ close (s);
+ return -1;
+ }
+ }
+
+ sin.sin_family = AF_INET;
+ memcpy (&(sin.sin_addr.s_addr), hp->h_addr, sizeof (hp->h_addr));
+ sin.sin_port = htons (server_port);
+
+ /* datagram socket */
+ if (connect (s, (struct sockaddr *) &sin, sizeof (sin)) < 0)
+ {
+ close (s);
+ return -1;
+ }
+
+ return s;
+}
+
+static int
+rtp_connect (char *hostname, int port)
+{
+ struct sockaddr_in sin;
+ struct timeval tv;
+ int err, err_len;
+ int rxsockbufsz;
+ int s;
+ fd_set set;
+
+ if (port <= 1023)
+ return -1;
+
+ s = socket (PF_INET, SOCK_DGRAM, 0);
+ if (s == -1)
+ return -1;
+
+ sin.sin_family = AF_INET;
+ if (!hostname || !strcmp (hostname, "0.0.0.0"))
+ sin.sin_addr.s_addr = htonl (INADDR_ANY);
+ else
+#ifndef HAVE_WINSOCK2
+#ifdef USE_ATON
+ inet_aton (hostname, &sin.sin_addr);
+#else
+ inet_pton (AF_INET, hostname, &sin.sin_addr);
+#endif
+#else
+ sin.sin_addr.s_addr = htonl (INADDR_ANY);
+#endif
+ sin.sin_port = htons (port);
+
+ /* Increase the socket rx buffer size to maximum -- this is UDP */
+ rxsockbufsz = 240 * 1024;
+ if (setsockopt (s, SOL_SOCKET, SO_RCVBUF,
+ &rxsockbufsz, sizeof (rxsockbufsz)))
+ mp_msg (MSGT_OPEN, MSGL_ERR, "Couldn't set receive socket buffer size\n");
+
+ /* if multicast address, add membership */
+ if ((ntohl (sin.sin_addr.s_addr) >> 28) == 0xe)
+ {
+ struct ip_mreq mcast;
+ mcast.imr_multiaddr.s_addr = sin.sin_addr.s_addr;
+ mcast.imr_interface.s_addr = 0;
+
+ if (setsockopt (s, IPPROTO_IP, IP_ADD_MEMBERSHIP, &mcast, sizeof (mcast)))
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "IP_ADD_MEMBERSHIP failed\n");
+ close (s);
+ return -1;
+ }
+ }
+
+ /* datagram socket */
+ if (bind (s, (struct sockaddr *) &sin, sizeof (sin)))
+ {
+#ifndef HAVE_WINSOCK2
+ if (errno != EINPROGRESS)
+#else
+ if (WSAGetLastError() != WSAEINPROGRESS)
+#endif
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "bind: %s\n", strerror (errno));
+ close (s);
+ return -1;
+ }
+ }
+
+ tv.tv_sec = 0;
+ tv.tv_usec = (1 * 1000000); /* 1 second timeout */
+
+ FD_ZERO (&set);
+ FD_SET (s, &set);
+
+ err = select (s + 1, &set, NULL, NULL, &tv);
+ if (err < 0)
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "Select failed: %s\n", strerror (errno));
+ close (s);
+ return -1;
+ }
+ else if (err == 0)
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "Timeout! No data from host %s\n", hostname);
+ close (s);
+ return -1;
+ }
+
+ err_len = sizeof (err);
+ getsockopt (s, SOL_SOCKET, SO_ERROR, &err, (socklen_t *) &err_len);
+ if (err)
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "Socket error: %d\n", err);
+ close (s);
+ return -1;
+ }
+
+ return s;
+}
+
+static int
+is_multicast_address (char *addr)
+{
+ struct sockaddr_in sin;
+
+ if (!addr)
+ return -1;
+
+ sin.sin_family = AF_INET;
+ inet_pton (AF_INET, addr, &sin.sin_addr);
+
+ if ((ntohl (sin.sin_addr.s_addr) >> 28) == 0xe)
+ return 1;
+
+ return 0;
+}
+
+struct rtp_rtsp_session_t *
+rtp_setup_and_play (rtsp_t *rtsp_session)
+{
+ struct rtp_rtsp_session_t* rtp_session = NULL;
+ const fsdp_media_description_t *med_dsc = NULL;
+ char temp_buf[MAX_LENGTH + 1];
+ char npt[256];
+
+ char* answer;
+ char* sdp;
+ char *server_addr = NULL;
+ char *destination = NULL;
+
+ int statut;
+ int content_length = 0;
+ int is_multicast = 0;
+
+ fsdp_description_t *dsc = NULL;
+ fsdp_error_t result;
+
+ int client_rtp_port = -1;
+ int client_rtcp_port = -1;
+ int server_rtp_port = -1;
+ int server_rtcp_port = -1;
+ int rtp_sock = -1;
+ int rtcp_sock = -1;
+
+ /* 1. send a RTSP DESCRIBE request to server */
+ rtsp_schedule_field (rtsp_session, RTSP_ACCEPT_SDP);
+ statut = rtsp_request_describe (rtsp_session, NULL);
+ if (statut < 200 || statut > 299)
+ return NULL;
+
+ answer = rtsp_search_answers (rtsp_session, RTSP_CONTENT_LENGTH);
+ if (answer)
+ content_length = atoi (answer);
+ else
+ return NULL;
+
+ answer = rtsp_search_answers (rtsp_session, RTSP_CONTENT_TYPE);
+ if (!answer || !strstr (answer, RTSP_APPLICATION_SDP))
+ return NULL;
+
+ /* 2. read SDP message from server */
+ sdp = (char *) malloc (content_length + 1);
+ if (rtsp_read_data (rtsp_session, sdp, content_length) <= 0)
+ {
+ free (sdp);
+ return NULL;
+ }
+ sdp[content_length] = 0;
+
+ /* 3. parse SDP message */
+ dsc = fsdp_description_new ();
+ result = fsdp_parse (sdp, dsc);
+ if (result != FSDPE_OK)
+ {
+ free (sdp);
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+ mp_msg (MSGT_OPEN, MSGL_V, "SDP:\n%s\n", sdp);
+ free (sdp);
+
+ /* 4. check for number of media streams: only one is supported */
+ if (fsdp_get_media_count (dsc) != 1)
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR,
+ "A single media stream only is supported atm.\n");
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* 5. set the Normal Play Time parameter
+ * use range provided by server in SDP or start now if empty */
+ sprintf (npt, RTSP_RANGE);
+ if (fsdp_get_range (dsc))
+ strcat (npt, fsdp_get_range (dsc));
+ else
+ strcat (npt, RTSP_NPT_NOW);
+
+ /* 5. check for a valid media stream */
+ med_dsc = fsdp_get_media (dsc, 0);
+ if (!med_dsc)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* 6. parse the `m=<media> <port> <transport> <fmt list>' line */
+
+ /* check for an A/V media */
+ if (fsdp_get_media_type (med_dsc) != FSDP_MEDIA_VIDEO &&
+ fsdp_get_media_type (med_dsc) != FSDP_MEDIA_AUDIO)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* only RTP/AVP transport method is supported right now */
+ if (fsdp_get_media_transport_protocol (med_dsc) != FSDP_TP_RTP_AVP)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* only MPEG-TS is supported at the moment */
+ if (!strstr (fsdp_get_media_format (med_dsc, 0),
+ RTSP_MEDIA_CONTAINER_MPEG_TS))
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* get client port (if any) advised by server */
+ client_rtp_port = fsdp_get_media_port (med_dsc);
+ if (client_rtp_port == -1)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* if client_rtp_port = 0 => let client randomly pick one */
+ if (client_rtp_port == 0)
+ {
+ /* TODO: we should check if the port is in use first */
+ if (rtsp_port)
+ client_rtp_port = rtsp_port;
+ else
+ client_rtp_port = RTSP_DEFAULT_PORT;
+ }
+
+ /* RTCP port generally is RTP port + 1 */
+ client_rtcp_port = client_rtp_port + 1;
+
+ mp_msg (MSGT_OPEN, MSGL_V,
+ "RTP Port from SDP appears to be: %d\n", client_rtp_port);
+ mp_msg (MSGT_OPEN, MSGL_V,
+ "RTCP Port from SDP appears to be: %d\n", client_rtcp_port);
+
+ /* 7. parse the `c=<network type> <addr type> <connection address>' line */
+
+ /* check for a valid media network type (inet) */
+ if (fsdp_get_media_network_type (med_dsc) != FSDP_NETWORK_TYPE_INET)
+ {
+ /* no control for media: try global one instead */
+ if (fsdp_get_global_conn_network_type (dsc) != FSDP_NETWORK_TYPE_INET)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+ }
+
+ /* only IPv4 is supported atm. */
+ if (fsdp_get_media_address_type (med_dsc) != FSDP_ADDRESS_TYPE_IPV4)
+ {
+ /* no control for media: try global one instead */
+ if (fsdp_get_global_conn_address_type (dsc) != FSDP_ADDRESS_TYPE_IPV4)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+ }
+
+ /* get the media server address to connect to */
+ if (fsdp_get_media_address (med_dsc))
+ server_addr = strdup (fsdp_get_media_address (med_dsc));
+ else if (fsdp_get_global_conn_address (dsc))
+ {
+ /* no control for media: try global one instead */
+ server_addr = strdup (fsdp_get_global_conn_address (dsc));
+ }
+
+ if (!server_addr)
+ {
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* check for a UNICAST or MULTICAST address to connect to */
+ is_multicast = is_multicast_address (server_addr);
+
+ /* 8. initiate an RTP session */
+ rtp_session = rtp_session_new ();
+ if (!rtp_session)
+ {
+ free (server_addr);
+ fsdp_description_delete (dsc);
+ return NULL;
+ }
+
+ /* get the media control URL */
+ if (fsdp_get_media_control (med_dsc, 0))
+ rtp_session->control_url = strdup (fsdp_get_media_control (med_dsc, 0));
+ fsdp_description_delete (dsc);
+ if (!rtp_session->control_url)
+ {
+ free (server_addr);
+ rtp_session_free (rtp_session);
+ return NULL;
+ }
+
+ /* 9. create the payload for RTSP SETUP request */
+ memset (temp_buf, '\0', MAX_LENGTH);
+ snprintf (temp_buf, MAX_LENGTH,
+ RTSP_TRANSPORT_REQUEST,
+ is_multicast ? RTSP_TRANSPORT_MULTICAST : RTSP_TRANSPORT_UNICAST,
+ is_multicast ? RTSP_MULTICAST_PORT : RTSP_UNICAST_CLIENT_PORT,
+ client_rtp_port, client_rtcp_port);
+ mp_msg (MSGT_OPEN, MSGL_V, "RTSP Transport: %s\n", temp_buf);
+
+ rtsp_unschedule_field (rtsp_session, RTSP_SESSION);
+ rtsp_schedule_field (rtsp_session, temp_buf);
+
+ /* 10. check for the media control URL type and initiate RTSP SETUP */
+ if (!strncmp (rtp_session->control_url, "rtsp://", 7)) /* absolute URL */
+ statut = rtsp_request_setup (rtsp_session,
+ rtp_session->control_url, NULL);
+ else /* relative URL */
+ statut = rtsp_request_setup (rtsp_session,
+ NULL, rtp_session->control_url);
+
+ if (statut < 200 || statut > 299)
+ {
+ free (server_addr);
+ rtp_session_free (rtp_session);
+ return NULL;
+ }
+
+ /* 11. parse RTSP SETUP response: we need it to actually determine
+ * the real address and port to connect to */
+ answer = rtsp_search_answers (rtsp_session, RTSP_TRANSPORT);
+ if (!answer)
+ {
+ free (server_addr);
+ rtp_session_free (rtp_session);
+ return NULL;
+ }
+
+ /* check for RTP and RTCP ports to bind according to how request was done */
+ is_multicast = 0;
+ if (strstr (answer, RTSP_TRANSPORT_MULTICAST))
+ is_multicast = 1;
+
+ if (is_multicast)
+ parse_port (answer, RTSP_MULTICAST_PORT,
+ &client_rtp_port, &client_rtcp_port);
+ else
+ {
+ parse_port (answer, RTSP_UNICAST_CLIENT_PORT,
+ &client_rtp_port, &client_rtcp_port);
+ parse_port (answer, RTSP_UNICAST_SERVER_PORT,
+ &server_rtp_port, &server_rtcp_port);
+ }
+
+ /* now check network settings as determined by server */
+ destination = parse_destination (answer);
+ if (!destination)
+ destination = strdup (server_addr);
+ free (server_addr);
+
+ mp_msg (MSGT_OPEN, MSGL_V, "RTSP Destination: %s\n", destination);
+ mp_msg (MSGT_OPEN, MSGL_V, "Client RTP port : %d\n", client_rtp_port);
+ mp_msg (MSGT_OPEN, MSGL_V, "Client RTCP port : %d\n", client_rtcp_port);
+ mp_msg (MSGT_OPEN, MSGL_V, "Server RTP port : %d\n", server_rtp_port);
+ mp_msg (MSGT_OPEN, MSGL_V, "Server RTCP port : %d\n", server_rtcp_port);
+
+ /* 12. performs RTSP PLAY request */
+ rtsp_schedule_field (rtsp_session, npt);
+ statut = rtsp_request_play (rtsp_session, NULL);
+ if (statut < 200 || statut > 299)
+ {
+ rtp_session_free (rtp_session);
+ return NULL;
+ }
+
+ /* 13. create RTP and RTCP connections */
+ rtp_sock = rtp_connect (destination, client_rtp_port);
+ rtcp_sock = rtcp_connect (client_rtcp_port, server_rtcp_port, destination);
+ rtp_session_set_fd (rtp_session, rtp_sock, rtcp_sock);
+ free (destination);
+
+ mp_msg (MSGT_OPEN, MSGL_V, "RTP Sock : %d\nRTCP Sock : %d\n",
+ rtp_session->rtp_socket, rtp_session->rtcp_socket);
+
+ if (rtp_session->rtp_socket == -1)
+ {
+ rtp_session_free (rtp_session);
+ return NULL;
+ }
+
+ return rtp_session;
+}
Added: trunk/libmpdemux/librtsp/rtsp_rtp.h
==============================================================================
--- (empty file)
+++ trunk/libmpdemux/librtsp/rtsp_rtp.h Mon Jun 26 23:27:57 2006
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2006 Benjamin Zores
+ * heavily base on the Freebox patch for xine by Vincent Mussard
+ * but with many enhancements for better RTSP RFC compliance.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef _HAVE_RTSP_RTP_H_
+#define _HAVE_RTSP_RTP_H_
+
+#include <pthread.h>
+
+#include "rtsp.h"
+
+#define MAX_PREVIEW_SIZE 4096
+
+struct rtp_rtsp_session_t {
+ int rtp_socket;
+ int rtcp_socket;
+ char *control_url;
+ int count;
+};
+
+struct rtp_rtsp_session_t *rtp_setup_and_play (rtsp_t* rtsp_session);
+off_t rtp_read (struct rtp_rtsp_session_t* st, char *buf, off_t length);
+void rtp_session_free (struct rtp_rtsp_session_t *st);
+void rtcp_send_rr (rtsp_t *s, struct rtp_rtsp_session_t *st);
+
+#endif /* _HAVE_RTSP_RTP_H_ */
+
Modified: trunk/libmpdemux/librtsp/rtsp_session.c
==============================================================================
--- trunk/libmpdemux/librtsp/rtsp_session.c (original)
+++ trunk/libmpdemux/librtsp/rtsp_session.c Mon Jun 26 23:27:57 2006
@@ -23,6 +23,9 @@
*
*
* high level interface to rtsp servers.
+ *
+ * 2006, Benjamin Zores and Vincent Mussard
+ * Support for MPEG-TS streaming through RFC compliant RTSP servers
*/
#include <sys/types.h>
@@ -42,7 +45,9 @@
#include <inttypes.h>
#include "mp_msg.h"
+#include "../rtp.h"
#include "rtsp.h"
+#include "rtsp_rtp.h"
#include "rtsp_session.h"
#include "../realrtsp/real.h"
#include "../realrtsp/rmff.h"
@@ -53,6 +58,7 @@
#define LOG
*/
+#define RTSP_OPTIONS_PUBLIC "Public"
#define RTSP_OPTIONS_SERVER "Server"
#define RTSP_OPTIONS_LOCATION "Location"
#define RTSP_OPTIONS_REAL "RealChallenge1"
@@ -63,6 +69,7 @@
struct rtsp_session_s {
rtsp_t *s;
struct real_rtsp_session_t* real_session;
+ struct rtp_rtsp_session_t* rtp_session;
};
//rtsp_session_t *rtsp_session_start(char *mrl) {
@@ -76,6 +83,7 @@
rtsp_session = malloc (sizeof (rtsp_session_t));
rtsp_session->s = NULL;
rtsp_session->real_session = NULL;
+ rtsp_session->rtp_session = NULL;
//connect:
*redir = 0;
@@ -141,13 +149,52 @@
rtsp_session->real_session->recv_size =
rtsp_session->real_session->header_len;
rtsp_session->real_session->recv_read = 0;
- } else
+ } else /* not a Real server : try RTP instead */
{
- mp_msg (MSGT_OPEN, MSGL_ERR,"rtsp_session: Not a Real server. Server type is '%s'.\n",server);
- rtsp_close(rtsp_session->s);
- free(server);
- free(rtsp_session);
- return NULL;
+ char *public = NULL;
+
+ /* look for the Public: field in response to RTSP OPTIONS */
+ public = strdup (rtsp_search_answers (rtsp_session->s,
+ RTSP_OPTIONS_PUBLIC));
+ if (!public)
+ {
+ rtsp_close (rtsp_session->s);
+ free (server);
+ free (mrl_line);
+ free (rtsp_session);
+ return NULL;
+ }
+
+ /* check for minimalistic RTSP RFC compliance */
+ if (!strstr (public, RTSP_METHOD_DESCRIBE)
+ || !strstr (public, RTSP_METHOD_SETUP)
+ || !strstr (public, RTSP_METHOD_PLAY)
+ || !strstr (public, RTSP_METHOD_TEARDOWN))
+ {
+ free (public);
+ mp_msg (MSGT_OPEN, MSGL_ERR,
+ "Remote server does not meet minimal RTSP 1.0 compliance.\n");
+ rtsp_close (rtsp_session->s);
+ free (server);
+ free (mrl_line);
+ free (rtsp_session);
+ return NULL;
+ }
+
+ free (public);
+ rtsp_session->rtp_session = rtp_setup_and_play (rtsp_session->s);
+
+ /* neither a Real or an RTP server */
+ if (!rtsp_session->rtp_session)
+ {
+ mp_msg (MSGT_OPEN, MSGL_ERR, "rtsp_session: unsupported RTSP server. ");
+ mp_msg (MSGT_OPEN, MSGL_ERR, "Server type is '%s'.\n", server);
+ rtsp_close (rtsp_session->s);
+ free (server);
+ free (mrl_line);
+ free (rtsp_session);
+ return NULL;
+ }
}
free(server);
@@ -194,6 +241,19 @@
return len;
}
+ else if (this->rtp_session)
+ {
+ int l = 0;
+
+ l = read_rtp_from_server (this->rtp_session->rtp_socket, data, len);
+ /* send RTSP and RTCP keepalive */
+ rtcp_send_rr (this->s, this->rtp_session);
+
+ if (l == 0)
+ rtsp_session_end (this);
+
+ return l;
+ }
return 0;
}
@@ -203,5 +263,7 @@
rtsp_close(session->s);
if (session->real_session)
free_real_rtsp_session (session->real_session);
+ if (session->rtp_session)
+ rtp_session_free (session->rtp_session);
free(session);
}
Modified: trunk/libmpdemux/librtsp/rtsp_session.h
==============================================================================
--- trunk/libmpdemux/librtsp/rtsp_session.h (original)
+++ trunk/libmpdemux/librtsp/rtsp_session.h Mon Jun 26 23:27:57 2006
@@ -23,6 +23,9 @@
*
*
* high level interface to rtsp servers.
+ *
+ * 2006, Benjamin Zores and Vincent Mussard
+ * Support for MPEG-TS streaming through RFC compliant RTSP servers
*/
#ifndef HAVE_RTSP_SESSION_H
Modified: trunk/libmpdemux/rtp.c
==============================================================================
--- trunk/libmpdemux/rtp.c (original)
+++ trunk/libmpdemux/rtp.c Mon Jun 26 23:27:57 2006
@@ -189,7 +189,7 @@
// Read next rtp packet using cache
-static int read_rtp_from_server(int fd, char *buffer, int length) {
+int read_rtp_from_server(int fd, char *buffer, int length) {
// Following test is ASSERT (i.e. uneuseful if code is correct)
if(buffer==NULL || length<STREAM_BUFFER_SIZE) {
mp_msg(MSGT_NETWORK, MSGL_ERR, "RTP buffer invalid; no data return from network\n");
Modified: trunk/libmpdemux/rtp.h
==============================================================================
--- trunk/libmpdemux/rtp.h (original)
+++ trunk/libmpdemux/rtp.h Mon Jun 26 23:27:57 2006
@@ -33,5 +33,6 @@
static int getrtp2(int fd, struct rtpheader *rh, char** data, int* lengthData);
+int read_rtp_from_server(int fd, char *buffer, int length);
#endif
More information about the MPlayer-cvslog
mailing list