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
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--- trunk/cfg-common.h (original) +++ trunk/cfg-common.h Mon Jun 26 23:27:57 2006 @@ -70,12 +70,11 @@ {"sdp", "-sdp is obsolete, use sdp://file instead.\n", CONF_TYPE_PRINT, 0, 0, 0, NULL}, // -rtsp-stream-over-tcp option, specifying TCP streaming of RTP/RTCP {"rtsp-stream-over-tcp",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
On Mon, 26 Jun 2006 23:27:58 +0200 (CEST) ben <subversion@mplayerhq.hu> wrote:
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)
For the record, Win32 code and ifdefs has been borrowed from libmpdemux/rtp.c. I have absolutely no idea whether it actually compiles/works or not and it could be really fine if someone can check this out (at least that it doesn't break compile). Some more piece of code might need portability against Win32, I'm really not aware at all of Winsock2 way of working. Ben -- "My life, and by extension everyone else's is meaningless." Bender, Futurama
On Monday, 26 June 2006 at 23:27, ben wrote:
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)
Does that mean live555 is no longer necessary? R. -- MPlayer developer and RPMs maintainer: http://rpm.greysector.net/mplayer/ There should be a science of discontent. People need hard times and oppression to develop psychic muscles. -- from "Collected Sayings of Muad'Dib" by the Princess Irulan
On Monday, 26 June 2006 at 23:27, ben wrote:
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)
Does that mean live555 is no longer necessary?
For watching MPEG-TS streams over RTSP/RTP yes. For any other content, LIVE555 still is needed. As stated before, RTSP demuxer is kinda generic but stream content is handled by RTP demuxer which has to support as many payloads as formats you want to support. Current implementation only support MPEG-TS so does my RTSP implementation. Expanding RTP demuxer to other payloads is required for RTSP demuxer to support them. Ben
Hi, On Mon, Jun 26, 2006 at 11:27:58PM +0200, ben wrote:
+#include <pthread.h>
Why is this here? I can't see it used anywhere. Anyway, it should have been under #ifdef HAVE_PTHREADS
+#include <netdb.h>
This belongs under the #ifndef as well. You can avoid these bugs by simply grepping for the headers you include and either use them in the same way or add a configure check.
Added: trunk/libmpdemux/librtsp/rtsp_rtp.h +#define _HAVE_RTSP_RTP_H_ + +#include <pthread.h>
Wow, that's the new definition of overkill.. including an unused header twice *g* Btw. I thought policy was to avoid including headers in headers? There is also a inet_pton somwhere without all the #ifndef HAVE_WINSOCK2 #ifdef USE_ATON stuff. And I thought "let's quickly test something here under windows...". IMHO if you can't get someone to test stuff, just disable it under that other OS. Greetings, Reimar Doeffinger
Hi, On Mon, Jun 26, 2006 at 11:27:58PM +0200, ben wrote:
+#include <pthread.h>
Why is this here? I can't see it used anywhere. Anyway, it should have been under #ifdef HAVE_PTHREADS
right, my previous code was using dedicated thread for rtp reader with ring buffer and was finally removed but i forgot about the header. I'll fix that.
There is also a inet_pton somwhere without all the #ifndef HAVE_WINSOCK2 #ifdef USE_ATON stuff.
will fix it, thx. Ben
Hi, On Tue, Jun 27, 2006 at 03:56:28PM +0200, Benjamin Zores wrote: [...]
There is also a inet_pton somwhere without all the #ifndef HAVE_WINSOCK2 #ifdef USE_ATON stuff.
will fix it, thx.
Oh, and all that #ifdef stuff won't work if you don't include config.h ... Greetings, Reimar Doeffinger
participants (4)
-
ben -
Benjamin Zores -
Dominik 'Rathann' Mierzejewski -
Reimar Doeffinger