comparison libmpdemux/network.c @ 12799:befb79b0232e

Make rtp:// cohexist with LIVE.COM Patch by Nico Sabbi

12798:33423862b159

#include "network.h"
#include "http.h"
#include "cookies.h"
#include "url.h"
#include "asf.h"
#ifndef STREAMING_LIVE_DOT_COM
#include "rtp.h"
#endif
#include "pnm.h"
#include "realrtsp/rtsp_session.h"

#include "../version.h"

	if( streaming_ctrl->buffer ) free( streaming_ctrl->buffer );
	if( streaming_ctrl->data ) free( streaming_ctrl->data );
	free( streaming_ctrl );
}

#ifndef STREAMING_LIVE_DOT_COM
int
read_rtp_from_server(int fd, char *buffer, int length) {
	struct rtpheader rh;
	char *data;
	int len;

	got_first = 1;
	sequence = rh.b.sequence;
	memcpy(buffer, data, len);
	return(len);
}
#endif


// Converts an address family constant to a string

char *af2String(int af) {

			mp_msg(MSGT_NETWORK,MSGL_ERR,"SIP support requires the \"LIVE.COM Streaming Media\" libraries!\n");
			return -1;
#endif
		}

#ifndef STREAMING_LIVE_DOT_COM
	// Old, hacked RTP support, which works for MPEG Program Streams
	//   RTP streams only:
		// Checking for RTP
		if( !strcasecmp(url->protocol, "rtp") ) {
			if( url->port==0 ) {
				mp_msg(MSGT_NETWORK,MSGL_ERR,"You must enter a port number for RTP streams!\n");
				return -1;
			}
			return 0;
		}
#endif

		// Checking for ASF
		if( !strncasecmp(url->protocol, "mms", 3) ) {
			*file_format = DEMUXER_TYPE_ASF;
			return 0;

	stream->streaming_ctrl->status = streaming_playing_e;
	return 0;
}


#ifndef STREAMING_LIVE_DOT_COM
// Start listening on a UDP port. If multicast, join the group.
int
rtp_open_socket( URL_t *url ) {
	int socket_server_fd, rxsockbufsz;
	int err, err_len;
	fd_set set;
	struct sockaddr_in server_address;

		}
	}
	return socket_server_fd;
}

int
rtp_streaming_read( int fd, char *buffer, int size, streaming_ctrl_t *streaming_ctrl ) {
    return read_rtp_from_server( fd, buffer, size );
}

int
rtp_streaming_start( stream_t *stream ) {
	streaming_ctrl_t *streaming_ctrl;
	int fd;

	if( stream==NULL ) return -1;

	streaming_ctrl->prebuffer_size = 64*1024;	// 64 KBytes	
	streaming_ctrl->buffering = 0;
	streaming_ctrl->status = streaming_playing_e;
	return 0;
}
#endif

int
streaming_start(stream_t *stream, int *demuxer_type, URL_t *url) {
	int ret;
	if( stream==NULL ) return -1;

	ret = -1;
	
	// Get the bandwidth available
	stream->streaming_ctrl->bandwidth = network_bandwidth;
	
#ifndef STREAMING_LIVE_DOT_COM
	// For RTP streams, we usually don't know the stream type until we open it.
	if( !strcasecmp( stream->streaming_ctrl->url->protocol, "rtp")) {
		if(stream->fd >= 0) {
			if(closesocket(stream->fd) < 0)
				mp_msg(MSGT_NETWORK,MSGL_ERR,"streaming_start : Closing socket %d failed %s\n",stream->fd,strerror(errno));
		}
		stream->fd = -1;
		ret = rtp_streaming_start( stream );
	} else
#endif

	if( !strcasecmp( stream->streaming_ctrl->url->protocol, "pnm")) {
		stream->fd = -1;
		ret = pnm_streaming_start( stream );
	} else

12799:befb79b0232e

#include "network.h"
#include "http.h"
#include "cookies.h"
#include "url.h"
#include "asf.h"
#include "rtp.h"
#include "pnm.h"
#include "realrtsp/rtsp_session.h"

#include "../version.h"

	if( streaming_ctrl->buffer ) free( streaming_ctrl->buffer );
	if( streaming_ctrl->data ) free( streaming_ctrl->data );
	free( streaming_ctrl );
}

int
read_rtp_from_server(int fd, char *buffer, int length) {
	struct rtpheader rh;
	char *data;
	int len;

	got_first = 1;
	sequence = rh.b.sequence;
	memcpy(buffer, data, len);
	return(len);
}


// Converts an address family constant to a string

char *af2String(int af) {

			mp_msg(MSGT_NETWORK,MSGL_ERR,"SIP support requires the \"LIVE.COM Streaming Media\" libraries!\n");
			return -1;
#endif
		}

	// Old, hacked RTP support, which works for MPEG Streams
	//   RTP streams only:
		// Checking for RTP
		if( !strcasecmp(url->protocol, "rtp") ) {
			if( url->port==0 ) {
				mp_msg(MSGT_NETWORK,MSGL_ERR,"You must enter a port number for RTP streams!\n");
				return -1;
			}
			return 0;
		}

		// Checking for ASF
		if( !strncasecmp(url->protocol, "mms", 3) ) {
			*file_format = DEMUXER_TYPE_ASF;
			return 0;

	stream->streaming_ctrl->status = streaming_playing_e;
	return 0;
}


// Start listening on a UDP port. If multicast, join the group.
static int
rtp_open_socket( URL_t *url ) {
	int socket_server_fd, rxsockbufsz;
	int err, err_len;
	fd_set set;
	struct sockaddr_in server_address;

		}
	}
	return socket_server_fd;
}

static int
rtp_streaming_read( int fd, char *buffer, int size, streaming_ctrl_t *streaming_ctrl ) {
    return read_rtp_from_server( fd, buffer, size );
}

static int
rtp_streaming_start( stream_t *stream ) {
	streaming_ctrl_t *streaming_ctrl;
	int fd;

	if( stream==NULL ) return -1;

	streaming_ctrl->prebuffer_size = 64*1024;	// 64 KBytes	
	streaming_ctrl->buffering = 0;
	streaming_ctrl->status = streaming_playing_e;
	return 0;
}

int
streaming_start(stream_t *stream, int *demuxer_type, URL_t *url) {
	int ret;
	if( stream==NULL ) return -1;

	ret = -1;
	
	// Get the bandwidth available
	stream->streaming_ctrl->bandwidth = network_bandwidth;
	
	// For RTP streams, we usually don't know the stream type until we open it.
	if( !strcasecmp( stream->streaming_ctrl->url->protocol, "rtp")) {
		if(stream->fd >= 0) {
			if(closesocket(stream->fd) < 0)
				mp_msg(MSGT_NETWORK,MSGL_ERR,"streaming_start : Closing socket %d failed %s\n",stream->fd,strerror(errno));
		}
		stream->fd = -1;
		ret = rtp_streaming_start( stream );
	} else

	if( !strcasecmp( stream->streaming_ctrl->url->protocol, "pnm")) {
		stream->fd = -1;
		ret = pnm_streaming_start( stream );
	} else