mirror of
https://github.com/mpv-player/mpv
synced 2024-12-16 20:05:07 +00:00
49867bd432
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@19277 b3059339-0415-0410-9bf9-f77b7e298cf2
701 lines
17 KiB
C
701 lines
17 KiB
C
/*
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* Copyright (C) 2006 Benjamin Zores
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* based on the Freebox patch for xine by Vincent Mussard
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* but with many enhancements for better RTSP RFC compliance.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <unistd.h>
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#include <stdio.h>
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#include <string.h>
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#include <errno.h>
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#include <stdlib.h>
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#include <sys/types.h>
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#include <inttypes.h>
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#include "config.h"
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#ifndef HAVE_WINSOCK2
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#include <netdb.h>
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#include <netinet/in.h>
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#include <sys/socket.h>
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#include <arpa/inet.h>
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#else
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#include <winsock2.h>
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#include <ws2tcpip.h>
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#endif
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#include "mp_msg.h"
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#include "rtsp.h"
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#include "rtsp_rtp.h"
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#include "rtsp_session.h"
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#include "../freesdp/common.h"
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#include "../freesdp/parser.h"
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#define RTSP_DEFAULT_PORT 31336
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#define MAX_LENGTH 256
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#define RTSP_ACCEPT_SDP "Accept: application/sdp"
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#define RTSP_CONTENT_LENGTH "Content-length"
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#define RTSP_CONTENT_TYPE "Content-Type"
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#define RTSP_APPLICATION_SDP "application/sdp"
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#define RTSP_RANGE "Range: "
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#define RTSP_NPT_NOW "npt=now-"
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#define RTSP_MEDIA_CONTAINER_MPEG_TS "33"
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#define RTSP_TRANSPORT_REQUEST "Transport: RTP/AVP;%s;%s%i-%i;mode=\"PLAY\""
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#define RTSP_TRANSPORT_MULTICAST "multicast"
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#define RTSP_TRANSPORT_UNICAST "unicast"
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#define RTSP_MULTICAST_PORT "port="
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#define RTSP_UNICAST_CLIENT_PORT "client_port="
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#define RTSP_UNICAST_SERVER_PORT "server_port="
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#define RTSP_SETUP_DESTINATION "destination="
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#define RTSP_SESSION "Session"
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#define RTSP_TRANSPORT "Transport"
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/* hardcoded RTCP RR - this is _NOT_ RFC compliant */
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#define RTCP_RR_SIZE 32
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#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"
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#define RTCP_SEND_FREQUENCY 1024
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int rtsp_port = 0;
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char *rtsp_destination = NULL;
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void
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rtcp_send_rr (rtsp_t *s, struct rtp_rtsp_session_t *st)
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{
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if (st->rtcp_socket == -1)
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return;
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/* send RTCP RR every RTCP_SEND_FREQUENCY packets
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* FIXME : NOT CORRECT, HARDCODED, BUT MAKES SOME SERVERS HAPPY
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* not rfc compliant
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* http://www.faqs.org/rfcs/rfc1889.html chapter 6 for RTCP
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*/
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if (st->count == RTCP_SEND_FREQUENCY)
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{
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char rtcp_content[RTCP_RR_SIZE];
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strcpy (rtcp_content, RTCP_RR);
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send (st->rtcp_socket, rtcp_content, RTCP_RR_SIZE, 0);
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/* ping RTSP server to keep connection alive.
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we use OPTIONS instead of PING as not all servers support it */
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rtsp_request_options (s, "*");
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st->count = 0;
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}
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else
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st->count++;
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}
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static struct rtp_rtsp_session_t *
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rtp_session_new (void)
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{
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struct rtp_rtsp_session_t *st = NULL;
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st = malloc (sizeof (struct rtp_rtsp_session_t));
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st->rtp_socket = -1;
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st->rtcp_socket = -1;
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st->control_url = NULL;
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st->count = 0;
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return st;
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}
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void
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rtp_session_free (struct rtp_rtsp_session_t *st)
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{
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if (!st)
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return;
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if (st->rtp_socket != -1)
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close (st->rtp_socket);
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if (st->rtcp_socket != -1)
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close (st->rtcp_socket);
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if (st->control_url)
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free (st->control_url);
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free (st);
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}
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static void
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rtp_session_set_fd (struct rtp_rtsp_session_t *st,
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int rtp_sock, int rtcp_sock)
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{
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if (!st)
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return;
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st->rtp_socket = rtp_sock;
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st->rtcp_socket = rtcp_sock;
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}
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static int
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parse_port (const char *line, const char *param,
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int *rtp_port, int *rtcp_port)
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{
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char *parse1;
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char *parse2;
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char *parse3;
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char *line_copy = strdup (line);
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parse1 = strstr (line_copy, param);
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if (parse1)
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{
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parse2 = strstr (parse1, "-");
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if (parse2)
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{
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parse3 = strstr (parse2, ";");
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if (parse3)
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parse3[0] = 0;
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parse2[0] = 0;
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}
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else
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{
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free (line_copy);
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return 0;
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}
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}
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else
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{
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free (line_copy);
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return 0;
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}
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*rtp_port = atoi (parse1 + strlen (param));
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*rtcp_port = atoi (parse2 + 1);
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free (line_copy);
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return 1;
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}
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static char *
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parse_destination (const char *line)
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{
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char *parse1;
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char *parse2;
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char *dest = NULL;
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char *line_copy = strdup (line);
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int len;
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parse1 = strstr (line_copy, RTSP_SETUP_DESTINATION);
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if (!parse1)
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{
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free (line_copy);
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return NULL;
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}
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parse2 = strstr (parse1, ";");
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if (!parse2)
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{
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free (line_copy);
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return NULL;
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}
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len = strlen (parse1) - strlen (parse2)
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- strlen (RTSP_SETUP_DESTINATION) + 1;
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dest = (char *) malloc (len + 1);
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snprintf (dest, len, parse1 + strlen (RTSP_SETUP_DESTINATION));
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free (line_copy);
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return dest;
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}
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static int
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rtcp_connect (int client_port, int server_port, const char* server_hostname)
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{
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struct sockaddr_in sin;
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struct hostent *hp;
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int s;
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if (client_port <= 1023)
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return -1;
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s = socket (PF_INET, SOCK_DGRAM, IPPROTO_UDP);
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if (s == -1)
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return -1;
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hp = gethostbyname (server_hostname);
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if (!hp)
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{
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close (s);
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return -1;
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}
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sin.sin_family = AF_INET;
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sin.sin_addr.s_addr = INADDR_ANY;
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sin.sin_port = htons (client_port);
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if (bind (s, (struct sockaddr *) &sin, sizeof (sin)))
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{
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#ifndef HAVE_WINSOCK2
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if (errno != EINPROGRESS)
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#else
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if (WSAGetLastError() != WSAEINPROGRESS)
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#endif
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{
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close (s);
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return -1;
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}
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}
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sin.sin_family = AF_INET;
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memcpy (&(sin.sin_addr.s_addr), hp->h_addr, sizeof (hp->h_addr));
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sin.sin_port = htons (server_port);
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/* datagram socket */
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if (connect (s, (struct sockaddr *) &sin, sizeof (sin)) < 0)
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{
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close (s);
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return -1;
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}
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return s;
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}
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static int
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rtp_connect (char *hostname, int port)
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{
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struct sockaddr_in sin;
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struct timeval tv;
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int err, err_len;
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int rxsockbufsz;
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int s;
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fd_set set;
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if (port <= 1023)
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return -1;
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s = socket (PF_INET, SOCK_DGRAM, 0);
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if (s == -1)
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return -1;
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sin.sin_family = AF_INET;
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if (!hostname || !strcmp (hostname, "0.0.0.0"))
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sin.sin_addr.s_addr = htonl (INADDR_ANY);
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else
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#ifndef HAVE_WINSOCK2
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#ifdef USE_ATON
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inet_aton (hostname, &sin.sin_addr);
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#else
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inet_pton (AF_INET, hostname, &sin.sin_addr);
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#endif
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#else
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sin.sin_addr.s_addr = htonl (INADDR_ANY);
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#endif
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sin.sin_port = htons (port);
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/* Increase the socket rx buffer size to maximum -- this is UDP */
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rxsockbufsz = 240 * 1024;
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if (setsockopt (s, SOL_SOCKET, SO_RCVBUF,
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&rxsockbufsz, sizeof (rxsockbufsz)))
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mp_msg (MSGT_OPEN, MSGL_ERR, "Couldn't set receive socket buffer size\n");
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/* if multicast address, add membership */
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if ((ntohl (sin.sin_addr.s_addr) >> 28) == 0xe)
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{
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struct ip_mreq mcast;
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mcast.imr_multiaddr.s_addr = sin.sin_addr.s_addr;
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mcast.imr_interface.s_addr = 0;
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if (setsockopt (s, IPPROTO_IP, IP_ADD_MEMBERSHIP, &mcast, sizeof (mcast)))
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{
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mp_msg (MSGT_OPEN, MSGL_ERR, "IP_ADD_MEMBERSHIP failed\n");
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close (s);
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return -1;
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}
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}
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/* datagram socket */
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if (bind (s, (struct sockaddr *) &sin, sizeof (sin)))
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{
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#ifndef HAVE_WINSOCK2
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if (errno != EINPROGRESS)
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#else
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if (WSAGetLastError() != WSAEINPROGRESS)
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#endif
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{
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mp_msg (MSGT_OPEN, MSGL_ERR, "bind: %s\n", strerror (errno));
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close (s);
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return -1;
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}
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}
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tv.tv_sec = 0;
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tv.tv_usec = (1 * 1000000); /* 1 second timeout */
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FD_ZERO (&set);
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FD_SET (s, &set);
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err = select (s + 1, &set, NULL, NULL, &tv);
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if (err < 0)
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{
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mp_msg (MSGT_OPEN, MSGL_ERR, "Select failed: %s\n", strerror (errno));
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close (s);
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return -1;
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}
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else if (err == 0)
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{
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mp_msg (MSGT_OPEN, MSGL_ERR, "Timeout! No data from host %s\n", hostname);
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close (s);
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return -1;
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}
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err_len = sizeof (err);
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getsockopt (s, SOL_SOCKET, SO_ERROR, &err, (socklen_t *) &err_len);
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if (err)
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{
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mp_msg (MSGT_OPEN, MSGL_ERR, "Socket error: %d\n", err);
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close (s);
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return -1;
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}
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return s;
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}
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static int
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is_multicast_address (char *addr)
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{
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struct sockaddr_in sin;
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if (!addr)
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return -1;
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sin.sin_family = AF_INET;
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#ifndef HAVE_WINSOCK2
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#ifdef USE_ATON
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inet_aton (addr, &sin.sin_addr);
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#else
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inet_pton (AF_INET, addr, &sin.sin_addr);
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#endif
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#else
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sin.sin_addr.s_addr = htonl (INADDR_ANY);
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#endif
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if ((ntohl (sin.sin_addr.s_addr) >> 28) == 0xe)
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return 1;
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return 0;
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}
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struct rtp_rtsp_session_t *
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rtp_setup_and_play (rtsp_t *rtsp_session)
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{
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struct rtp_rtsp_session_t* rtp_session = NULL;
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const fsdp_media_description_t *med_dsc = NULL;
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char temp_buf[MAX_LENGTH + 1];
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char npt[256];
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char* answer;
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char* sdp;
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char *server_addr = NULL;
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char *destination = NULL;
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int statut;
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int content_length = 0;
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int is_multicast = 0;
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fsdp_description_t *dsc = NULL;
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fsdp_error_t result;
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int client_rtp_port = -1;
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int client_rtcp_port = -1;
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int server_rtp_port = -1;
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int server_rtcp_port = -1;
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int rtp_sock = -1;
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int rtcp_sock = -1;
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/* 1. send a RTSP DESCRIBE request to server */
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rtsp_schedule_field (rtsp_session, RTSP_ACCEPT_SDP);
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statut = rtsp_request_describe (rtsp_session, NULL);
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if (statut < 200 || statut > 299)
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return NULL;
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answer = rtsp_search_answers (rtsp_session, RTSP_CONTENT_LENGTH);
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if (answer)
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content_length = atoi (answer);
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else
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return NULL;
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answer = rtsp_search_answers (rtsp_session, RTSP_CONTENT_TYPE);
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if (!answer || !strstr (answer, RTSP_APPLICATION_SDP))
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return NULL;
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/* 2. read SDP message from server */
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sdp = (char *) malloc (content_length + 1);
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if (rtsp_read_data (rtsp_session, sdp, content_length) <= 0)
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{
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free (sdp);
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return NULL;
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}
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sdp[content_length] = 0;
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/* 3. parse SDP message */
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dsc = fsdp_description_new ();
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result = fsdp_parse (sdp, dsc);
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if (result != FSDPE_OK)
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{
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free (sdp);
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fsdp_description_delete (dsc);
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return NULL;
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}
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mp_msg (MSGT_OPEN, MSGL_V, "SDP:\n%s\n", sdp);
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free (sdp);
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/* 4. check for number of media streams: only one is supported */
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if (fsdp_get_media_count (dsc) != 1)
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{
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mp_msg (MSGT_OPEN, MSGL_ERR,
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"A single media stream only is supported atm.\n");
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* 5. set the Normal Play Time parameter
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* use range provided by server in SDP or start now if empty */
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sprintf (npt, RTSP_RANGE);
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if (fsdp_get_range (dsc))
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strcat (npt, fsdp_get_range (dsc));
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else
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strcat (npt, RTSP_NPT_NOW);
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/* 5. check for a valid media stream */
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med_dsc = fsdp_get_media (dsc, 0);
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if (!med_dsc)
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{
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* 6. parse the `m=<media> <port> <transport> <fmt list>' line */
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/* check for an A/V media */
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if (fsdp_get_media_type (med_dsc) != FSDP_MEDIA_VIDEO &&
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fsdp_get_media_type (med_dsc) != FSDP_MEDIA_AUDIO)
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{
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* only RTP/AVP transport method is supported right now */
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if (fsdp_get_media_transport_protocol (med_dsc) != FSDP_TP_RTP_AVP)
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{
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* only MPEG-TS is supported at the moment */
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if (!strstr (fsdp_get_media_format (med_dsc, 0),
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RTSP_MEDIA_CONTAINER_MPEG_TS))
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{
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* get client port (if any) advised by server */
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client_rtp_port = fsdp_get_media_port (med_dsc);
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if (client_rtp_port == -1)
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{
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fsdp_description_delete (dsc);
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return NULL;
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}
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/* if client_rtp_port = 0 => let client randomly pick one */
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if (client_rtp_port == 0)
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{
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/* TODO: we should check if the port is in use first */
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if (rtsp_port)
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client_rtp_port = rtsp_port;
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else
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client_rtp_port = RTSP_DEFAULT_PORT;
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}
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/* 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 */
|
|
if (rtsp_destination)
|
|
destination = strdup (rtsp_destination);
|
|
else
|
|
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)
|
|
{
|
|
free (destination);
|
|
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;
|
|
}
|