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* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
* This library is distributed in the hope that it will be useful,
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
#define HAVE_AV_CONFIG_H
#include "avformat.h"
#include <stdarg.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/poll.h>
#include <errno.h>
#include <sys/time.h>
#include <time.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#ifdef CONFIG_HAVE_DLFCN
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#include <dlfcn.h>
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#include "ffserver.h"
/* maximum number of simultaneous HTTP connections */
#define HTTP_MAX_CONNECTIONS 2000
enum HTTPState {
HTTPSTATE_WAIT_REQUEST,
HTTPSTATE_SEND_HEADER,
HTTPSTATE_SEND_DATA_HEADER,
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HTTPSTATE_SEND_DATA, /* sending TCP or UDP data */
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HTTPSTATE_RECEIVE_DATA,
HTTPSTATE_WAIT_FEED, /* wait for data from the feed */
HTTPSTATE_WAIT, /* wait before sending next packets */
HTTPSTATE_WAIT_SHORT, /* short wait for short term
bandwidth limitation */
HTTPSTATE_READY,
RTSPSTATE_WAIT_REQUEST,
RTSPSTATE_SEND_REPLY,
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"HTTP_WAIT_REQUEST",
"HTTP_SEND_HEADER",
"SEND_DATA_HEADER",
"SEND_DATA",
"SEND_DATA_TRAILER",
"RECEIVE_DATA",
"WAIT_FEED",
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"WAIT",
"WAIT_SHORT",
"READY",
"RTSP_WAIT_REQUEST",
"RTSP_SEND_REPLY",
/* coef for exponential mean for bitrate estimation in statistics */
#define AVG_COEF 0.9
/* timeouts are in ms */
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#define HTTP_REQUEST_TIMEOUT (15 * 1000)
#define RTSP_REQUEST_TIMEOUT (3600 * 24 * 1000)
typedef struct {
INT64 count1, count2;
long time1, time2;
} DataRateData;
/* context associated with one connection */
typedef struct HTTPContext {
enum HTTPState state;
int fd; /* socket file descriptor */
struct sockaddr_in from_addr; /* origin */
struct pollfd *poll_entry; /* used when polling */
long timeout;
UINT8 *buffer_ptr, *buffer_end;
int http_error;
struct HTTPContext *next;
int got_key_frame; /* stream 0 => 1, stream 1 => 2, stream 2=> 4 */
INT64 data_count;
/* feed input */
int feed_fd;
/* input format handling */
AVFormatContext *fmt_in;
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long start_time; /* In milliseconds - this wraps fairly often */
INT64 first_pts; /* initial pts value */
int pts_stream_index; /* stream we choose as clock reference */
/* -1 is invalid stream */
int feed_streams[MAX_STREAMS]; /* index of streams in the feed */
int switch_feed_streams[MAX_STREAMS]; /* index of streams in the feed */
int switch_pending;
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AVFormatContext fmt_ctx; /* instance of FFStream for one user */
int suppress_log;
int wmp_client_id;
char protocol[16];
char method[16];
char url[128];
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int is_packetized; /* if true, the stream is packetized */
int packet_stream_index; /* current stream for output in state machine */
/* RTSP state specific */
UINT8 *pb_buffer; /* XXX: use that in all the code */
ByteIOContext *pb;
int seq; /* RTSP sequence number */
/* RTP state specific */
enum RTSPProtocol rtp_protocol;
char session_id[32]; /* session id */
AVFormatContext *rtp_ctx[MAX_STREAMS];
URLContext *rtp_handles[MAX_STREAMS];
/* RTP short term bandwidth limitation */
int packet_byte_count;
int packet_start_time_us; /* used for short durations (a few
seconds max) */
} HTTPContext;
/* each generated stream is described here */
enum StreamType {
STREAM_TYPE_LIVE,
STREAM_TYPE_STATUS,
enum IPAddressAction {
IP_ALLOW = 1,
IP_DENY,
};
typedef struct IPAddressACL {
struct IPAddressACL *next;
enum IPAddressAction action;
struct in_addr first;
struct in_addr last;
} IPAddressACL;
/* description of each stream of the ffserver.conf file */
typedef struct FFStream {
enum StreamType stream_type;
char filename[1024]; /* stream filename */
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struct FFStream *feed; /* feed we are using (can be null if
coming from file) */
IPAddressACL *acl;
int prebuffer; /* Number of millseconds early to start */
long max_time; /* Number of milliseconds to run */
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int send_on_key;
AVStream *streams[MAX_STREAMS];
int feed_streams[MAX_STREAMS]; /* index of streams in the feed */
char feed_filename[1024]; /* file name of the feed storage, or
input file name for a stream */
char author[512];
char title[512];
char copyright[512];
char comment[512];
time_t pid_start; /* Of ffmpeg process */
int bandwidth; /* bandwidth, in kbits/s */
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/* RTSP options */
char *rtsp_option;
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/* multicast specific */
int is_multicast;
struct in_addr multicast_ip;
int multicast_port; /* first port used for multicast */
int multicast_ttl;
int loop; /* if true, send the stream in loops (only meaningful if file) */
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int feed_opened; /* true if someone is writing to the feed */
int conns_served;
INT64 bytes_served;
INT64 feed_max_size; /* maximum storage size */
INT64 feed_write_index; /* current write position in feed (it wraps round) */
INT64 feed_size; /* current size of feed */
struct FFStream *next_feed;
} FFStream;
typedef struct FeedData {
long long data_count;
float avg_frame_size; /* frame size averraged over last frames with exponential mean */
} FeedData;
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struct sockaddr_in my_http_addr;
struct sockaddr_in my_rtsp_addr;
char logfilename[1024];
HTTPContext *first_http_ctx;
FFStream *first_feed; /* contains only feeds */
FFStream *first_stream; /* contains all streams, including feeds */
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static void new_connection(int server_fd, int is_rtsp);
static void close_connection(HTTPContext *c);
/* HTTP handling */
static int handle_connection(HTTPContext *c);
static int http_send_data(HTTPContext *c);
static void compute_stats(HTTPContext *c);
static int open_input_stream(HTTPContext *c, const char *info);
static int http_start_receive_data(HTTPContext *c);
static int http_receive_data(HTTPContext *c);
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static int compute_send_delay(HTTPContext *c);
/* RTSP handling */
static int rtsp_parse_request(HTTPContext *c);
static void rtsp_cmd_describe(HTTPContext *c, const char *url);
static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h);
static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h);
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/* SDP handling */
static int prepare_sdp_description(FFStream *stream, UINT8 **pbuffer,
struct in_addr my_ip);
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/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
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FFStream *stream, const char *session_id);
static int rtp_new_av_stream(HTTPContext *c,
int stream_index, struct sockaddr_in *dest_addr);
static const char *my_program_dir;
static int ffserver_debug;
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static int ffserver_daemon;
static int no_launch;
static int need_to_start_children;
int max_bandwidth;
int current_bandwidth;
static long cur_time; // Making this global saves on passing it around everywhere
static long gettime_ms(void)
{
struct timeval tv;
gettimeofday(&tv,NULL);
return (long long)tv.tv_sec * 1000 + (tv.tv_usec / 1000);
}
static FILE *logfile = NULL;
static void http_log(char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
if (logfile) {
fflush(logfile);
}
static char *ctime1(char *buf2)
{
time_t ti;
char *p;
ti = time(NULL);
p = ctime(&ti);
strcpy(buf2, p);
p = buf2 + strlen(p) - 1;
if (*p == '\n')
*p = '\0';
return buf2;
}
static void log_connection(HTTPContext *c)
{
char buf2[32];
if (c->suppress_log)
return;
http_log("%s - - [%s] \"%s %s %s\" %d %lld\n",
inet_ntoa(c->from_addr.sin_addr),
ctime1(buf2), c->method, c->url,
c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
static void update_datarate(DataRateData *drd, INT64 count)
{
if (!drd->time1 && !drd->count1) {
drd->time1 = drd->time2 = cur_time;
drd->count1 = drd->count2 = count;
} else {
if (cur_time - drd->time2 > 5000) {
drd->time1 = drd->time2;
drd->count1 = drd->count2;
drd->time2 = cur_time;
drd->count2 = count;
}
}
}
/* In bytes per second */
static int compute_datarate(DataRateData *drd, INT64 count)
{
if (cur_time == drd->time1)
return 0;
return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
}
static int get_longterm_datarate(DataRateData *drd, INT64 count)
{
/* You get the first 3 seconds flat out */
if (cur_time - drd->time1 < 3000)
return 0;
return compute_datarate(drd, count);
}
static void start_children(FFStream *feed)
{
if (no_launch)
return;
if (feed->child_argv && !feed->pid) {
feed->pid_start = time(0);
feed->pid = fork();
if (feed->pid < 0) {
fprintf(stderr, "Unable to create children\n");
exit(1);
}
if (!feed->pid) {
/* In child */
char pathname[1024];
char *slash;
int i;
for (i = 3; i < 256; i++) {
close(i);
}
i = open("/dev/null", O_RDWR);
if (i)
dup2(i, 0);
dup2(i, 1);
dup2(i, 2);
pstrcpy(pathname, sizeof(pathname), my_program_name);
slash = strrchr(pathname, '/');
if (!slash) {
slash = pathname;
} else {
slash++;
}
strcpy(slash, "ffmpeg");
/* This is needed to make relative pathnames work */
chdir(my_program_dir);
execvp(pathname, feed->child_argv);
_exit(1);
}
}
}
}
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/* open a listening socket */
static int socket_open_listen(struct sockaddr_in *my_addr)
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int server_fd, tmp;
server_fd = socket(AF_INET,SOCK_STREAM,0);
if (server_fd < 0) {
perror ("socket");
return -1;
}
tmp = 1;
setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
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if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
perror ("bind");
close(server_fd);
return -1;
}
if (listen (server_fd, 5) < 0) {
perror ("listen");
close(server_fd);
return -1;
}
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fcntl(server_fd, F_SETFL, O_NONBLOCK);
return server_fd;
}
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/* start all multicast streams */
static void start_multicast(void)
{
FFStream *stream;
char session_id[32];
HTTPContext *rtp_c;
struct sockaddr_in dest_addr;
int default_port, stream_index;
default_port = 6000;
for(stream = first_stream; stream != NULL; stream = stream->next) {
if (stream->is_multicast) {
/* open the RTP connection */
snprintf(session_id, sizeof(session_id),
"%08x%08x", (int)random(), (int)random());
/* choose a port if none given */
if (stream->multicast_port == 0) {
stream->multicast_port = default_port;
default_port += 100;
}
dest_addr.sin_family = AF_INET;
dest_addr.sin_addr = stream->multicast_ip;
dest_addr.sin_port = htons(stream->multicast_port);
rtp_c = rtp_new_connection(&dest_addr, stream, session_id);
if (!rtp_c) {
continue;
}
if (open_input_stream(rtp_c, "") < 0) {
fprintf(stderr, "Could not open input stream for stream '%s'\n",
stream->filename);
continue;
}
rtp_c->rtp_protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
/* open each RTP stream */
for(stream_index = 0; stream_index < stream->nb_streams;
stream_index++) {
dest_addr.sin_port = htons(stream->multicast_port +
2 * stream_index);
if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
stream->filename, stream_index);
exit(1);
}
}
/* change state to send data */
rtp_c->state = HTTPSTATE_SEND_DATA;
}
}
}
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/* main loop of the http server */
static int http_server(void)
{
int server_fd, ret, rtsp_server_fd, delay, delay1;
struct pollfd poll_table[HTTP_MAX_CONNECTIONS + 2], *poll_entry;
HTTPContext *c, *c_next;
server_fd = socket_open_listen(&my_http_addr);
if (server_fd < 0)
return -1;
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rtsp_server_fd = socket_open_listen(&my_rtsp_addr);
if (rtsp_server_fd < 0)
return -1;
first_http_ctx = NULL;
nb_connections = 0;
first_http_ctx = NULL;
start_multicast();
for(;;) {
poll_entry = poll_table;
poll_entry->fd = server_fd;
poll_entry->events = POLLIN;
poll_entry++;
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poll_entry->fd = rtsp_server_fd;
poll_entry->events = POLLIN;
poll_entry++;
/* wait for events on each HTTP handle */
c = first_http_ctx;
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delay = 1000;
while (c != NULL) {
int fd;
fd = c->fd;
switch(c->state) {
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case HTTPSTATE_SEND_HEADER:
case RTSPSTATE_SEND_REPLY:
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poll_entry->events = POLLOUT;
poll_entry++;
break;
case HTTPSTATE_SEND_DATA_HEADER:
case HTTPSTATE_SEND_DATA:
case HTTPSTATE_SEND_DATA_TRAILER:
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if (!c->is_packetized) {
/* for TCP, we output as much as we can (may need to put a limit) */
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLOUT;
poll_entry++;
} else {
/* not strictly correct, but currently cannot add
more than one fd in poll entry */
delay = 0;
}
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case HTTPSTATE_WAIT_REQUEST:
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case RTSPSTATE_WAIT_REQUEST:
/* need to catch errors */
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLIN;/* Maybe this will work */
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case HTTPSTATE_WAIT:
c->poll_entry = NULL;
delay1 = compute_send_delay(c);
if (delay1 < delay)
delay = delay1;
break;
case HTTPSTATE_WAIT_SHORT:
c->poll_entry = NULL;
delay1 = 10; /* one tick wait XXX: 10 ms assumed */
if (delay1 < delay)
delay = delay1;
break;
default:
c->poll_entry = NULL;
break;
}
c = c->next;
}
/* wait for an event on one connection. We poll at least every
second to handle timeouts */
do {
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ret = poll(poll_table, poll_entry - poll_table, delay);
if (need_to_start_children) {
need_to_start_children = 0;
start_children(first_feed);
}
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for(c = first_http_ctx; c != NULL; c = c_next) {
c_next = c->next;
if (handle_connection(c) < 0) {
log_connection(c);
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close_connection(c);
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/* new HTTP connection request ? */
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new_connection(server_fd, 0);
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/* new RTSP connection request ? */
if (poll_entry->revents & POLLIN) {
new_connection(rtsp_server_fd, 1);
}
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/* start waiting for a new HTTP/RTSP request */
static void start_wait_request(HTTPContext *c, int is_rtsp)
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c->buffer_ptr = c->buffer;
c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */
if (is_rtsp) {
c->timeout = cur_time + RTSP_REQUEST_TIMEOUT;
c->state = RTSPSTATE_WAIT_REQUEST;
} else {
c->timeout = cur_time + HTTP_REQUEST_TIMEOUT;
c->state = HTTPSTATE_WAIT_REQUEST;
}
}
static void new_connection(int server_fd, int is_rtsp)
{
struct sockaddr_in from_addr;
int fd, len;
HTTPContext *c = NULL;
len = sizeof(from_addr);
fd = accept(server_fd, (struct sockaddr *)&from_addr,
&len);
if (fd < 0)
return;
fcntl(fd, F_SETFL, O_NONBLOCK);
/* XXX: should output a warning page when coming
close to the connection limit */
if (nb_connections >= nb_max_connections)
goto fail;
/* add a new connection */
c = av_mallocz(sizeof(HTTPContext));
if (!c)
goto fail;
c->next = first_http_ctx;
first_http_ctx = c;
c->fd = fd;
c->poll_entry = NULL;
c->from_addr = from_addr;
c->buffer_size = IOBUFFER_INIT_SIZE;
c->buffer = av_malloc(c->buffer_size);
if (!c->buffer)
goto fail;
nb_connections++;
start_wait_request(c, is_rtsp);
return;
fail:
if (c) {
av_free(c->buffer);
av_free(c);
}
close(fd);
}
static void close_connection(HTTPContext *c)
{
HTTPContext **cp, *c1;
int i, nb_streams;
AVFormatContext *ctx;
URLContext *h;
AVStream *st;
/* remove connection from list */
cp = &first_http_ctx;
while ((*cp) != NULL) {
c1 = *cp;
if (c1 == c) {
*cp = c->next;
} else {
cp = &c1->next;
}
}
/* remove connection associated resources */
if (c->fd >= 0)
close(c->fd);
if (c->fmt_in) {
/* close each frame parser */
for(i=0;i<c->fmt_in->nb_streams;i++) {
st = c->fmt_in->streams[i];
if (st->codec.codec) {
avcodec_close(&st->codec);
}
}
av_close_input_file(c->fmt_in);
}
/* free RTP output streams if any */
nb_streams = 0;
if (c->stream)
nb_streams = c->stream->nb_streams;
for(i=0;i<nb_streams;i++) {
ctx = c->rtp_ctx[i];
if (ctx) {
av_write_trailer(ctx);
av_free(ctx);
}
h = c->rtp_handles[i];
if (h) {
url_close(h);
}
}
if (c->stream)
current_bandwidth -= c->stream->bandwidth;
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av_freep(&c->pb_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
}
static int handle_connection(HTTPContext *c)
{
int len, ret;
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case RTSPSTATE_WAIT_REQUEST:
/* timeout ? */
if ((c->timeout - cur_time) < 0)
return -1;
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
/* no need to read if no events */
if (!(c->poll_entry->revents & POLLIN))
return 0;
/* read the data */
len = read(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR)
return -1;
} else if (len == 0) {
return -1;
} else {
/* search for end of request. XXX: not fully correct since garbage could come after the end */
UINT8 *ptr;
c->buffer_ptr += len;
ptr = c->buffer_ptr;
if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
(ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
/* request found : parse it and reply */
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if (c->state == HTTPSTATE_WAIT_REQUEST) {
ret = http_parse_request(c);
} else {
ret = rtsp_parse_request(c);
}
if (ret < 0)
return -1;
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
}
}
break;
case HTTPSTATE_SEND_HEADER:
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
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/* no need to write if no events */
if (!(c->poll_entry->revents & POLLOUT))
return 0;
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
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av_freep(&c->pb_buffer);
if (c->stream)
c->stream->bytes_served += len;
c->data_count += len;
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av_freep(&c->pb_buffer);
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if (c->http_error) {
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}
/* all the buffer was sent : synchronize to the incoming stream */
c->state = HTTPSTATE_SEND_DATA_HEADER;
c->buffer_ptr = c->buffer_end = c->buffer;
}
}
break;
case HTTPSTATE_SEND_DATA:
case HTTPSTATE_SEND_DATA_HEADER:
case HTTPSTATE_SEND_DATA_TRAILER:
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/* for packetized output, we consider we can always write (the
input streams sets the speed). It may be better to verify
that we do not rely too much on the kernel queues */
if (!c->is_packetized) {
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
/* no need to read if no events */
if (!(c->poll_entry->revents & POLLOUT))
return 0;
}
if (http_send_data(c) < 0)
return -1;
break;
case HTTPSTATE_RECEIVE_DATA:
/* no need to read if no events */
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
if (!(c->poll_entry->revents & POLLIN))
return 0;
if (http_receive_data(c) < 0)
return -1;
break;
case HTTPSTATE_WAIT_FEED:
/* no need to read if no events */
if (c->poll_entry->revents & (POLLIN | POLLERR | POLLHUP))
return -1;
/* nothing to do, we'll be waken up by incoming feed packets */
break;
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case HTTPSTATE_WAIT:
/* if the delay expired, we can send new packets */
if (compute_send_delay(c) <= 0)
c->state = HTTPSTATE_SEND_DATA;
break;
case HTTPSTATE_WAIT_SHORT:
/* just return back to send data */
c->state = HTTPSTATE_SEND_DATA;
break;
case RTSPSTATE_SEND_REPLY:
if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
av_freep(&c->pb_buffer);
return -1;
}
/* no need to write if no events */
if (!(c->poll_entry->revents & POLLOUT))
return 0;
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
av_freep(&c->pb_buffer);
return -1;
}
} else {
c->buffer_ptr += len;
c->data_count += len;
if (c->buffer_ptr >= c->buffer_end) {
/* all the buffer was sent : wait for a new request */
av_freep(&c->pb_buffer);
start_wait_request(c, 1);
}
}
break;
case HTTPSTATE_READY:
/* nothing to do */
break;
static int extract_rates(char *rates, int ratelen, const char *request)
{
const char *p;
for (p = request; *p && *p != '\r' && *p != '\n'; ) {
if (strncasecmp(p, "Pragma:", 7) == 0) {
const char *q = p + 7;
while (*q && *q != '\n' && isspace(*q))
q++;
if (strncasecmp(q, "stream-switch-entry=", 20) == 0) {
int stream_no;
int rate_no;
q += 20;
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while (1) {
while (*q && *q != '\n' && *q != ':')
q++;
if (sscanf(q, ":%d:%d", &stream_no, &rate_no) != 2) {
break;
}
stream_no--;
if (stream_no < ratelen && stream_no >= 0) {
rates[stream_no] = rate_no;
}
while (*q && *q != '\n' && !isspace(*q))
q++;
}
return 1;
}
}
p = strchr(p, '\n');
if (!p)
break;
p++;
}
return 0;
}
static int find_stream_in_feed(FFStream *feed, AVCodecContext *codec, int bit_rate)
{
int i;
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int best_bitrate = 100000000;
int best = -1;
for (i = 0; i < feed->nb_streams; i++) {
AVCodecContext *feed_codec = &feed->streams[i]->codec;
if (feed_codec->codec_id != codec->codec_id ||
feed_codec->sample_rate != codec->sample_rate ||
feed_codec->width != codec->width ||
feed_codec->height != codec->height) {
continue;
}
/* Potential stream */
/* We want the fastest stream less than bit_rate, or the slowest
* faster than bit_rate
*/
if (feed_codec->bit_rate <= bit_rate) {
if (best_bitrate > bit_rate || feed_codec->bit_rate > best_bitrate) {
best_bitrate = feed_codec->bit_rate;
best = i;
}
} else {
if (feed_codec->bit_rate < best_bitrate) {
best_bitrate = feed_codec->bit_rate;
best = i;
}
}
}
return best;
}
static int modify_current_stream(HTTPContext *c, char *rates)
{
int i;
FFStream *req = c->stream;
int action_required = 0;
for (i = 0; i < req->nb_streams; i++) {
AVCodecContext *codec = &req->streams[i]->codec;
switch(rates[i]) {
case 0:
c->switch_feed_streams[i] = req->feed_streams[i];
break;
case 1:
c->switch_feed_streams[i] = find_stream_in_feed(req->feed, codec, codec->bit_rate / 2);
break;
case 2:
/* Wants off or slow */
c->switch_feed_streams[i] = find_stream_in_feed(req->feed, codec, codec->bit_rate / 4);
#ifdef WANTS_OFF
/* This doesn't work well when it turns off the only stream! */
c->switch_feed_streams[i] = -2;
c->feed_streams[i] = -2;
#endif
break;
}
if (c->switch_feed_streams[i] >= 0 && c->switch_feed_streams[i] != c->feed_streams[i])
action_required = 1;
}
static void do_switch_stream(HTTPContext *c, int i)
{
if (c->switch_feed_streams[i] >= 0) {
#ifdef PHILIP
c->feed_streams[i] = c->switch_feed_streams[i];
#endif
}
}
Fabrice Bellard
committed
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/* XXX: factorize in utils.c ? */
/* XXX: take care with different space meaning */
static void skip_spaces(const char **pp)
{
const char *p;
p = *pp;
while (*p == ' ' || *p == '\t')
p++;
*pp = p;
}
static void get_word(char *buf, int buf_size, const char **pp)
{
const char *p;
char *q;
p = *pp;
skip_spaces(&p);
q = buf;
while (!isspace(*p) && *p != '\0') {
if ((q - buf) < buf_size - 1)
*q++ = *p;
p++;
}
if (buf_size > 0)
*q = '\0';
*pp = p;
}
static int validate_acl(FFStream *stream, HTTPContext *c)
{
enum IPAddressAction last_action = IP_DENY;
IPAddressACL *acl;
struct in_addr *src = &c->from_addr.sin_addr;
for (acl = stream->acl; acl; acl = acl->next) {
if (src->s_addr >= acl->first.s_addr && src->s_addr <= acl->last.s_addr) {
return (acl->action == IP_ALLOW) ? 1 : 0;
}
last_action = acl->action;
}
/* Nothing matched, so return not the last action */
return (last_action == IP_DENY) ? 1 : 0;
}
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committed
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/* compute the real filename of a file by matching it without its
extensions to all the stream filenames */
static void compute_real_filename(char *filename, int max_size)
{
char file1[1024];
char file2[1024];
char *p;
FFStream *stream;
/* compute filename by matching without the file extensions */
pstrcpy(file1, sizeof(file1), filename);
p = strrchr(file1, '.');
if (p)
*p = '\0';
for(stream = first_stream; stream != NULL; stream = stream->next) {
pstrcpy(file2, sizeof(file2), stream->filename);
p = strrchr(file2, '.');
if (p)
*p = '\0';
if (!strcmp(file1, file2)) {
pstrcpy(filename, max_size, stream->filename);
break;
}
}
}
enum RedirType {
REDIR_NONE,
REDIR_ASX,
REDIR_RAM,
REDIR_ASF,
REDIR_RTSP,
REDIR_SDP,
};
/* parse http request and prepare header */
static int http_parse_request(HTTPContext *c)
{
char *p;
int post;
Fabrice Bellard
committed
enum RedirType redir_type;
char cmd[32];
char info[1024], *filename;
char url[1024], *q;
char protocol[32];
char msg[1024];
const char *mime_type;
FFStream *stream;
char ratebuf[32];
Fabrice Bellard
committed
get_word(cmd, sizeof(cmd), (const char **)&p);
pstrcpy(c->method, sizeof(c->method), cmd);
if (!strcmp(cmd, "GET"))
post = 0;
else if (!strcmp(cmd, "POST"))
post = 1;
else
return -1;
Fabrice Bellard
committed
get_word(url, sizeof(url), (const char **)&p);
pstrcpy(c->url, sizeof(c->url), url);
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committed
get_word(protocol, sizeof(protocol), (const char **)&p);
if (strcmp(protocol, "HTTP/1.0") && strcmp(protocol, "HTTP/1.1"))
return -1;
pstrcpy(c->protocol, sizeof(c->protocol), protocol);
/* find the filename and the optional info string in the request */
p = url;
if (*p == '/')
p++;
filename = p;
p = strchr(p, '?');
if (p) {
pstrcpy(info, sizeof(info), p);
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
if (strncasecmp(p, "User-Agent:", 11) == 0) {
useragent = p + 11;
if (*useragent && *useragent != '\n' && isspace(*useragent))
useragent++;
break;
}
p = strchr(p, '\n');
if (!p)
break;
p++;
}
Fabrice Bellard
committed
redir_type = REDIR_NONE;
if (match_ext(filename, "asx")) {
redir_type = REDIR_ASX;
filename[strlen(filename)-1] = 'f';
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committed
} else if (match_ext(filename, ".asf") &&
(!useragent || strncasecmp(useragent, "NSPlayer", 8) != 0)) {
/* if this isn't WMP or lookalike, return the redirector file */
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committed
redir_type = REDIR_ASF;
} else if (match_ext(filename, "rpm,ram")) {
redir_type = REDIR_RAM;
strcpy(filename + strlen(filename)-2, "m");
Fabrice Bellard
committed
} else if (match_ext(filename, "rtsp")) {
redir_type = REDIR_RTSP;
compute_real_filename(filename, sizeof(url) - 1);
} else if (match_ext(filename, "sdp")) {
redir_type = REDIR_SDP;
compute_real_filename(filename, sizeof(url) - 1);
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committed
if (!strcmp(stream->filename, filename) && validate_acl(stream, c))
break;
stream = stream->next;
}
if (stream == NULL) {
sprintf(msg, "File '%s' not found", url);
goto send_error;
}
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c->stream = stream;
memcpy(c->feed_streams, stream->feed_streams, sizeof(c->feed_streams));
memset(c->switch_feed_streams, -1, sizeof(c->switch_feed_streams));
if (stream->stream_type == STREAM_TYPE_REDIRECT) {
c->http_error = 301;
q = c->buffer;
q += sprintf(q, "HTTP/1.0 301 Moved\r\n");
q += sprintf(q, "Location: %s\r\n", stream->feed_filename);
q += sprintf(q, "Content-type: text/html\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "<html><head><title>Moved</title></head><body>\r\n");
q += sprintf(q, "You should be <a href=\"%s\">redirected</a>.\r\n", stream->feed_filename);
q += sprintf(q, "</body></html>\r\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
c->buffer_end = q;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
/* If this is WMP, get the rate information */
if (extract_rates(ratebuf, sizeof(ratebuf), c->buffer)) {
if (modify_current_stream(c, ratebuf)) {
for (i = 0; i < sizeof(c->feed_streams) / sizeof(c->feed_streams[0]); i++) {
if (c->switch_feed_streams[i] >= 0)
do_switch_stream(c, i);
}
}
}
if (post == 0 && stream->stream_type == STREAM_TYPE_LIVE) {
current_bandwidth += stream->bandwidth;
if (post == 0 && max_bandwidth < current_bandwidth) {
c->http_error = 200;
q = c->buffer;
q += sprintf(q, "HTTP/1.0 200 Server too busy\r\n");
q += sprintf(q, "Content-type: text/html\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "<html><head><title>Too busy</title></head><body>\r\n");
q += sprintf(q, "The server is too busy to serve your request at this time.<p>\r\n");
q += sprintf(q, "The bandwidth being served (including your stream) is %dkbit/sec, and this exceeds the limit of %dkbit/sec\r\n",
current_bandwidth, max_bandwidth);
q += sprintf(q, "</body></html>\r\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
c->buffer_end = q;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
Fabrice Bellard
committed
if (redir_type != REDIR_NONE) {
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char *hostinfo = 0;
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
if (strncasecmp(p, "Host:", 5) == 0) {
hostinfo = p + 5;
break;
}
p = strchr(p, '\n');
if (!p)
break;
p++;
}
if (hostinfo) {
char *eoh;
char hostbuf[260];
while (isspace(*hostinfo))
hostinfo++;
eoh = strchr(hostinfo, '\n');
if (eoh) {
if (eoh[-1] == '\r')
eoh--;
if (eoh - hostinfo < sizeof(hostbuf) - 1) {
memcpy(hostbuf, hostinfo, eoh - hostinfo);
hostbuf[eoh - hostinfo] = 0;
c->http_error = 200;
q = c->buffer;
Fabrice Bellard
committed
switch(redir_type) {
case REDIR_ASX:
q += sprintf(q, "HTTP/1.0 200 ASX Follows\r\n");
q += sprintf(q, "Content-type: video/x-ms-asf\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "<ASX Version=\"3\">\r\n");
q += sprintf(q, "<!-- Autogenerated by ffserver -->\r\n");
q += sprintf(q, "<ENTRY><REF HREF=\"http://%s/%s%s\"/></ENTRY>\r\n",
hostbuf, filename, info);
q += sprintf(q, "</ASX>\r\n");
Fabrice Bellard
committed
break;
case REDIR_RAM:
q += sprintf(q, "HTTP/1.0 200 RAM Follows\r\n");
q += sprintf(q, "Content-type: audio/x-pn-realaudio\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "# Autogenerated by ffserver\r\n");
q += sprintf(q, "http://%s/%s%s\r\n",
hostbuf, filename, info);
Fabrice Bellard
committed
break;
case REDIR_ASF:
q += sprintf(q, "HTTP/1.0 200 ASF Redirect follows\r\n");
q += sprintf(q, "Content-type: video/x-ms-asf\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "[Reference]\r\n");
q += sprintf(q, "Ref1=http://%s/%s%s\r\n",
hostbuf, filename, info);
Fabrice Bellard
committed
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break;
case REDIR_RTSP:
{
char hostname[256], *p;
/* extract only hostname */
pstrcpy(hostname, sizeof(hostname), hostbuf);
p = strrchr(hostname, ':');
if (p)
*p = '\0';
q += sprintf(q, "HTTP/1.0 200 RTSP Redirect follows\r\n");
/* XXX: incorrect mime type ? */
q += sprintf(q, "Content-type: application/x-rtsp\r\n");
q += sprintf(q, "\r\n");
q += sprintf(q, "rtsp://%s:%d/%s\r\n",
hostname, ntohs(my_rtsp_addr.sin_port),
filename);
}
break;
case REDIR_SDP:
{
UINT8 *sdp_data;
int sdp_data_size, len;
struct sockaddr_in my_addr;
q += sprintf(q, "HTTP/1.0 200 OK\r\n");
q += sprintf(q, "Content-type: application/sdp\r\n");
q += sprintf(q, "\r\n");
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
/* XXX: should use a dynamic buffer */
sdp_data_size = prepare_sdp_description(stream,
&sdp_data,
my_addr.sin_addr);
if (sdp_data_size > 0) {
memcpy(q, sdp_data, sdp_data_size);
q += sdp_data_size;
*q = '\0';
av_free(sdp_data);
}
}
break;
default:
av_abort();
Fabrice Bellard
committed
break;
Fabrice Bellard
committed
}
/* prepare output buffer */
c->buffer_ptr = c->buffer;
c->buffer_end = q;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
}
}
sprintf(msg, "ASX/RAM file not handled");
goto send_error;
stream->conns_served++;
/* XXX: add there authenticate and IP match */
if (post) {
/* if post, it means a feed is being sent */
if (!stream->is_feed) {
/* However it might be a status report from WMP! Lets log the data
* as it might come in handy one day
*/
char *logline = 0;
int client_id = 0;
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
if (strncasecmp(p, "Pragma: log-line=", 17) == 0) {
logline = p;
break;
}
if (strncasecmp(p, "Pragma: client-id=", 18) == 0) {
client_id = strtol(p + 18, 0, 10);
}
p = strchr(p, '\n');
if (!p)
break;
p++;
}
if (logline) {
char *eol = strchr(logline, '\n');
logline += 17;
if (eol) {
if (eol[-1] == '\r')
eol--;
http_log("%.*s\n", eol - logline, logline);
c->suppress_log = 1;
}
}
#ifdef DEBUG_WMP
http_log("\nGot request:\n%s\n", c->buffer);
#endif
if (client_id && extract_rates(ratebuf, sizeof(ratebuf), c->buffer)) {
HTTPContext *wmpc;
/* Now we have to find the client_id */
for (wmpc = first_http_ctx; wmpc; wmpc = wmpc->next) {
if (wmpc->wmp_client_id == client_id)
break;
}
if (wmpc) {
if (modify_current_stream(wmpc, ratebuf)) {
wmpc->switch_pending = 1;
}
}
}
sprintf(msg, "POST command not handled");
goto send_error;
}
if (http_start_receive_data(c) < 0) {
sprintf(msg, "could not open feed");
goto send_error;
}
c->http_error = 0;
c->state = HTTPSTATE_RECEIVE_DATA;
return 0;
}
if (strcmp(stream->filename + strlen(stream->filename) - 4, ".asf") == 0) {
http_log("\nGot request:\n%s\n", c->buffer);
}
#endif
if (c->stream->stream_type == STREAM_TYPE_STATUS)
goto send_stats;
/* open input stream */
if (open_input_stream(c, info) < 0) {
sprintf(msg, "Input stream corresponding to '%s' not found", url);
goto send_error;
}
/* prepare http header */
q = c->buffer;
q += sprintf(q, "HTTP/1.0 200 OK\r\n");
mime_type = c->stream->fmt->mime_type;
if (!mime_type)
mime_type = "application/x-octet_stream";
q += sprintf(q, "Pragma: no-cache\r\n");
/* for asf, we need extra headers */
if (!strcmp(c->stream->fmt->name,"asf_stream")) {
/* Need to allocate a client id */
c->wmp_client_id = random() & 0x7fffffff;
q += sprintf(q, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id);
Philip Gladstone
committed
q += sprintf(q, "Content-Type: %s\r\n", mime_type);
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q += sprintf(q, "\r\n");
/* prepare output buffer */
c->http_error = 0;
c->buffer_ptr = c->buffer;
c->buffer_end = q;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
send_error:
c->http_error = 404;
q = c->buffer;
q += sprintf(q, "HTTP/1.0 404 Not Found\r\n");
q += sprintf(q, "Content-type: %s\r\n", "text/html");
q += sprintf(q, "\r\n");
q += sprintf(q, "<HTML>\n");
q += sprintf(q, "<HEAD><TITLE>404 Not Found</TITLE></HEAD>\n");
q += sprintf(q, "<BODY>%s</BODY>\n", msg);
q += sprintf(q, "</HTML>\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
c->buffer_end = q;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
send_stats:
compute_stats(c);
c->http_error = 200; /* horrible : we use this value to avoid
going to the send data state */
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
Fabrice Bellard
committed
static void fmt_bytecount(ByteIOContext *pb, INT64 count)
{
static const char *suffix = " kMGTP";
const char *s;
for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++) {
}
Fabrice Bellard
committed
url_fprintf(pb, "%lld%c", count, *s);
}
static void compute_stats(HTTPContext *c)
{
HTTPContext *c1;
FFStream *stream;
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committed
char *p;
Fabrice Bellard
committed
int i, len;
ByteIOContext pb1, *pb = &pb1;
Fabrice Bellard
committed
if (url_open_dyn_buf(pb) < 0) {
/* XXX: return an error ? */
Fabrice Bellard
committed
c->buffer_end = c->buffer;
return;
Fabrice Bellard
committed
url_fprintf(pb, "HTTP/1.0 200 OK\r\n");
url_fprintf(pb, "Content-type: %s\r\n", "text/html");
url_fprintf(pb, "Pragma: no-cache\r\n");
url_fprintf(pb, "\r\n");
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committed
url_fprintf(pb, "<HEAD><TITLE>FFServer Status</TITLE>\n");
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url_fprintf(pb, "<link rel=\"shortcut icon\" href=\"%s\">\n", c->stream->feed_filename);
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url_fprintf(pb, "</HEAD>\n<BODY>");
url_fprintf(pb, "<H1>FFServer Status</H1>\n");
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url_fprintf(pb, "<H2>Available Streams</H2>\n");
url_fprintf(pb, "<TABLE cellspacing=0 cellpadding=4>\n");
url_fprintf(pb, "<TR><Th valign=top>Path<th align=left>Served<br>Conns<Th><br>bytes<Th valign=top>Format<Th>Bit rate<br>kbits/s<Th align=left>Video<br>kbits/s<th><br>Codec<Th align=left>Audio<br>kbits/s<th><br>Codec<Th align=left valign=top>Feed\n");
char sfilename[1024];
char *eosf;
if (stream->feed != stream) {
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pstrcpy(sfilename, sizeof(sfilename) - 10, stream->filename);
eosf = sfilename + strlen(sfilename);
if (eosf - sfilename >= 4) {
if (strcmp(eosf - 4, ".asf") == 0) {
strcpy(eosf - 4, ".asx");
} else if (strcmp(eosf - 3, ".rm") == 0) {
strcpy(eosf - 3, ".ram");
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} else if (stream->fmt == &rtp_mux) {
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/* generate a sample RTSP director if
unicast. Generate an SDP redirector if
multicast */
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eosf = strrchr(sfilename, '.');
if (!eosf)
eosf = sfilename + strlen(sfilename);
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if (stream->is_multicast)
strcpy(eosf, ".sdp");
else
strcpy(eosf, ".rtsp");
}
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url_fprintf(pb, "<TR><TD><A HREF=\"/%s\">%s</A> ",
sfilename, stream->filename);
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url_fprintf(pb, "<td align=right> %d <td align=right> ",
stream->conns_served);
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fmt_bytecount(pb, stream->bytes_served);
switch(stream->stream_type) {
case STREAM_TYPE_LIVE:
{
int audio_bit_rate = 0;
int video_bit_rate = 0;
const char *audio_codec_name = "";
const char *video_codec_name = "";
const char *audio_codec_name_extra = "";
const char *video_codec_name_extra = "";
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for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
AVCodec *codec = avcodec_find_encoder(st->codec.codec_id);
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
audio_bit_rate += st->codec.bit_rate;
if (codec) {
if (*audio_codec_name)
audio_codec_name_extra = "...";
audio_codec_name = codec->name;
}
break;
case CODEC_TYPE_VIDEO:
video_bit_rate += st->codec.bit_rate;
if (codec) {
if (*video_codec_name)
video_codec_name_extra = "...";
video_codec_name = codec->name;
}
break;
default:
av_abort();
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}
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url_fprintf(pb, "<TD align=center> %s <TD align=right> %d <TD align=right> %d <TD> %s %s <TD align=right> %d <TD> %s %s",
stream->fmt->name,
stream->bandwidth,
video_bit_rate / 1000, video_codec_name, video_codec_name_extra,
audio_bit_rate / 1000, audio_codec_name, audio_codec_name_extra);
if (stream->feed) {
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url_fprintf(pb, "<TD>%s", stream->feed->filename);
} else {
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url_fprintf(pb, "<TD>%s", stream->feed_filename);
}
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url_fprintf(pb, "\n");
break;
default:
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url_fprintf(pb, "<TD align=center> - <TD align=right> - <TD align=right> - <td><td align=right> - <TD>\n");
break;
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url_fprintf(pb, "</TABLE>\n");
stream = first_stream;
while (stream != NULL) {
if (stream->feed == stream) {
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url_fprintf(pb, "<h2>Feed %s</h2>", stream->filename);
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url_fprintf(pb, "Running as pid %d.\n", stream->pid);
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#if defined(linux) && !defined(CONFIG_NOCUTILS)
{
FILE *pid_stat;
char ps_cmd[64];
/* This is somewhat linux specific I guess */
snprintf(ps_cmd, sizeof(ps_cmd),
"ps -o \"%%cpu,cputime\" --no-headers %d",
stream->pid);
pid_stat = popen(ps_cmd, "r");
if (pid_stat) {
char cpuperc[10];
char cpuused[64];
if (fscanf(pid_stat, "%10s %64s", cpuperc,
cpuused) == 2) {
url_fprintf(pb, "Currently using %s%% of the cpu. Total time used %s.\n",
cpuperc, cpuused);
}
fclose(pid_stat);
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url_fprintf(pb, "<p>");
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url_fprintf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>type<th>kbits/s<th align=left>codec<th align=left>Parameters\n");
for (i = 0; i < stream->nb_streams; i++) {
AVStream *st = stream->streams[i];
AVCodec *codec = avcodec_find_encoder(st->codec.codec_id);
char *type = "unknown";
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char parameters[64];
parameters[0] = 0;
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
type = "audio";
break;
case CODEC_TYPE_VIDEO:
type = "video";
sprintf(parameters, "%dx%d, q=%d-%d, fps=%d", st->codec.width, st->codec.height,
st->codec.qmin, st->codec.qmax, st->codec.frame_rate / FRAME_RATE_BASE);
break;
default:
av_abort();
}
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url_fprintf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d<td>%s<td>%s\n",
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i, type, st->codec.bit_rate/1000, codec ? codec->name : "", parameters);
}
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url_fprintf(pb, "</table>\n");
}
stream = stream->next;
}
#if 0
{
float avg;
AVCodecContext *enc;
char buf[1024];
/* feed status */
stream = first_feed;
while (stream != NULL) {
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url_fprintf(pb, "<H1>Feed '%s'</H1>\n", stream->filename);
url_fprintf(pb, "<TABLE>\n");
url_fprintf(pb, "<TR><TD>Parameters<TD>Frame count<TD>Size<TD>Avg bitrate (kbits/s)\n");
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
FeedData *fdata = st->priv_data;
enc = &st->codec;
avcodec_string(buf, sizeof(buf), enc);
avg = fdata->avg_frame_size * (float)enc->rate * 8.0;
if (enc->codec->type == CODEC_TYPE_AUDIO && enc->frame_size > 0)
avg /= enc->frame_size;
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url_fprintf(pb, "<TR><TD>%s <TD> %d <TD> %Ld <TD> %0.1f\n",
buf, enc->frame_number, fdata->data_count, avg / 1000.0);
}
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url_fprintf(pb, "</TABLE>\n");
stream = stream->next_feed;
}
}
#endif
/* connection status */
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url_fprintf(pb, "<H2>Connection Status</H2>\n");
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url_fprintf(pb, "Number of connections: %d / %d<BR>\n",
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url_fprintf(pb, "Bandwidth in use: %dk / %dk<BR>\n",
current_bandwidth, max_bandwidth);
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url_fprintf(pb, "<TABLE>\n");
url_fprintf(pb, "<TR><th>#<th>File<th>IP<th>Proto<th>State<th>Target bits/sec<th>Actual bits/sec<th>Bytes transferred\n");
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while (c1 != NULL) {
int bitrate;
int j;
bitrate = 0;
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if (c1->stream) {
for (j = 0; j < c1->stream->nb_streams; j++) {
if (!c1->stream->feed) {
bitrate += c1->stream->streams[j]->codec.bit_rate;
} else {
if (c1->feed_streams[j] >= 0) {
bitrate += c1->stream->feed->streams[c1->feed_streams[j]]->codec.bit_rate;
}
}
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url_fprintf(pb, "<TR><TD><B>%d</B><TD>%s%s<TD>%s<TD>%s<TD>%s<td align=right>",
i,
c1->stream ? c1->stream->filename : "",
c1->state == HTTPSTATE_RECEIVE_DATA ? "(input)" : "",
p,
c1->protocol,
http_state[c1->state]);
fmt_bytecount(pb, bitrate);
url_fprintf(pb, "<td align=right>");
fmt_bytecount(pb, compute_datarate(&c1->datarate, c1->data_count) * 8);
url_fprintf(pb, "<td align=right>");
fmt_bytecount(pb, c1->data_count);
url_fprintf(pb, "\n");
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url_fprintf(pb, "</TABLE>\n");
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url_fprintf(pb, "<HR size=1 noshade>Generated at %s", p);
url_fprintf(pb, "</BODY>\n</HTML>\n");
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len = url_close_dyn_buf(pb, &c->pb_buffer);
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
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/* check if the parser needs to be opened for stream i */
static void open_parser(AVFormatContext *s, int i)
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AVStream *st = s->streams[i];
AVCodec *codec;
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committed
if (!st->codec.codec) {
codec = avcodec_find_decoder(st->codec.codec_id);
if (codec && (codec->capabilities & CODEC_CAP_PARSE_ONLY)) {
st->codec.parse_only = 1;
if (avcodec_open(&st->codec, codec) < 0) {
st->codec.parse_only = 0;
}
}
static int open_input_stream(HTTPContext *c, const char *info)
{
char buf[128];
char input_filename[1024];
AVFormatContext *s;
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int buf_size, i;
INT64 stream_pos;
/* find file name */
if (c->stream->feed) {
strcpy(input_filename, c->stream->feed->feed_filename);
buf_size = FFM_PACKET_SIZE;
/* compute position (absolute time) */
if (find_info_tag(buf, sizeof(buf), "date", info)) {
stream_pos = parse_date(buf, 0);
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committed
} else if (find_info_tag(buf, sizeof(buf), "buffer", info)) {
int prebuffer = strtol(buf, 0, 10);
stream_pos = av_gettime() - prebuffer * (INT64)1000000;
stream_pos = av_gettime() - c->stream->prebuffer * (INT64)1000;
}
} else {
strcpy(input_filename, c->stream->feed_filename);
buf_size = 0;
/* compute position (relative time) */
if (find_info_tag(buf, sizeof(buf), "date", info)) {
stream_pos = parse_date(buf, 1);
} else {
stream_pos = 0;
}
}
if (input_filename[0] == '\0')
return -1;
#if 0
{ time_t when = stream_pos / 1000000;
http_log("Stream pos = %lld, time=%s", stream_pos, ctime(&when));
}
#endif
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if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 0) {
http_log("%s not found", input_filename);
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}
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/* open each parser */
for(i=0;i<s->nb_streams;i++)
open_parser(s, i);
/* choose stream as clock source (we favorize video stream if
present) for packet sending */
c->pts_stream_index = 0;
for(i=0;i<c->stream->nb_streams;i++) {
if (c->pts_stream_index == 0 &&
c->stream->streams[i]->codec.codec_type == CODEC_TYPE_VIDEO) {
c->pts_stream_index = i;
}
}
if (c->fmt_in->iformat->read_seek) {
c->fmt_in->iformat->read_seek(c->fmt_in, stream_pos);
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/* set the start time (needed for maxtime and RTP packet timing) */
c->start_time = cur_time;
c->first_pts = AV_NOPTS_VALUE;
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/* currently desactivated because the new PTS handling is not
satisfactory yet */
//#define AV_READ_FRAME
#ifdef AV_READ_FRAME
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/* XXX: generalize that in ffmpeg for picture/audio/data. Currently
the return packet MUST NOT be freed */
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
{
AVStream *st;
int len, ret, old_nb_streams, i;
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/* see if remaining frames must be parsed */
for(;;) {
if (s->cur_len > 0) {
st = s->streams[s->cur_pkt.stream_index];
len = avcodec_parse_frame(&st->codec, &pkt->data, &pkt->size,
s->cur_ptr, s->cur_len);
if (len < 0) {
/* error: get next packet */
s->cur_len = 0;
} else {
s->cur_ptr += len;
s->cur_len -= len;
if (pkt->size) {
/* init pts counter if not done */
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
st->pts_incr = (INT64)s->pts_den;
av_frac_init(&st->pts, st->pts.val, 0,
(INT64)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
st->pts_incr = (INT64)s->pts_den * FRAME_RATE_BASE;
av_frac_init(&st->pts, st->pts.val, 0,
(INT64)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
}
}
/* a frame was read: return it */
pkt->pts = st->pts.val;
#if 0
printf("add pts=%Lx num=%Lx den=%Lx incr=%Lx\n",
st->pts.val, st->pts.num, st->pts.den, st->pts_incr);
#endif
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
av_frac_add(&st->pts, st->pts_incr * st->codec.frame_size);
break;
case CODEC_TYPE_VIDEO:
av_frac_add(&st->pts, st->pts_incr);
break;
default:
av_abort();
}
pkt->stream_index = s->cur_pkt.stream_index;
/* we use the codec indication because it is
more accurate than the demux flags */
pkt->flags = 0;
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committed
pkt->flags |= PKT_FLAG_KEY;
return 0;
}
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/* free previous packet */
av_free_packet(&s->cur_pkt);
old_nb_streams = s->nb_streams;
ret = av_read_packet(s, &s->cur_pkt);
if (ret)
return ret;
/* open parsers for each new streams */
for(i = old_nb_streams; i < s->nb_streams; i++)
open_parser(s, i);
st = s->streams[s->cur_pkt.stream_index];
/* update current pts (XXX: dts handling) from packet, or
use current pts if none given */
if (s->cur_pkt.pts != AV_NOPTS_VALUE) {
av_frac_set(&st->pts, s->cur_pkt.pts);
} else {
s->cur_pkt.pts = st->pts.val;
}
if (!st->codec.codec) {
/* no codec opened: just return the raw packet */
*pkt = s->cur_pkt;
/* no codec opened: just update the pts by considering we
have one frame and free the packet */
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
st->pts_incr = (INT64)s->pts_den * st->codec.frame_size;
av_frac_init(&st->pts, st->pts.val, 0,
(INT64)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
st->pts_incr = (INT64)s->pts_den * FRAME_RATE_BASE;
av_frac_init(&st->pts, st->pts.val, 0,
(INT64)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
}
}
av_frac_add(&st->pts, st->pts_incr);
return 0;
} else {
s->cur_ptr = s->cur_pkt.data;
s->cur_len = s->cur_pkt.size;
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2049
}
}
static int compute_send_delay(HTTPContext *c)
{
INT64 cur_pts, delta_pts, next_pts;
int delay1;
/* compute current pts value from system time */
cur_pts = ((INT64)(cur_time - c->start_time) * c->fmt_in->pts_den) /
(c->fmt_in->pts_num * 1000LL);
/* compute the delta from the stream we choose as
main clock (we do that to avoid using explicit
buffers to do exact packet reordering for each
stream */
/* XXX: really need to fix the number of streams */
if (c->pts_stream_index >= c->fmt_in->nb_streams)
next_pts = cur_pts;
else
next_pts = c->fmt_in->streams[c->pts_stream_index]->pts.val;
delta_pts = next_pts - cur_pts;
if (delta_pts <= 0) {
delay1 = 0;
} else {
delay1 = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
}
return delay1;
}
#else
/* just fall backs */
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
{
return av_read_packet(s, pkt);
}
static int compute_send_delay(HTTPContext *c)
{
int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
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return 0;
}
#endif
static int http_prepare_data(HTTPContext *c)
{
int i, len, ret;
AVFormatContext *ctx;
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
pstrcpy(c->fmt_ctx.author, sizeof(c->fmt_ctx.author),
c->stream->author);
pstrcpy(c->fmt_ctx.comment, sizeof(c->fmt_ctx.comment),
c->stream->comment);
pstrcpy(c->fmt_ctx.copyright, sizeof(c->fmt_ctx.copyright),
c->stream->copyright);
pstrcpy(c->fmt_ctx.title, sizeof(c->fmt_ctx.title),
c->stream->title);
/* open output stream by using specified codecs */
c->fmt_ctx.oformat = c->stream->fmt;
c->fmt_ctx.nb_streams = c->stream->nb_streams;
for(i=0;i<c->fmt_ctx.nb_streams;i++) {
AVStream *st;
st = av_mallocz(sizeof(AVStream));
c->fmt_ctx.streams[i] = st;
/* if file or feed, then just take streams from FFStream struct */
if (!c->stream->feed ||
c->stream->feed == c->stream)
memcpy(st, c->stream->streams[i], sizeof(AVStream));
else
memcpy(st, c->stream->feed->streams[c->stream->feed_streams[i]],
sizeof(AVStream));
st->codec.frame_number = 0; /* XXX: should be done in
AVStream, not in codec */
}
c->got_key_frame = 0;
/* prepare header and save header data in a stream */
if (url_open_dyn_buf(&c->fmt_ctx.pb) < 0) {
/* XXX: potential leak */
return -1;
}
c->fmt_ctx.pb.is_streamed = 1;
av_write_header(&c->fmt_ctx);
len = url_close_dyn_buf(&c->fmt_ctx.pb, &c->pb_buffer);
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
c->state = HTTPSTATE_SEND_DATA;
c->last_packet_sent = 0;
break;
case HTTPSTATE_SEND_DATA:
/* find a new packet */
{
AVPacket pkt;
Fabrice Bellard
committed
/* read a packet from the input stream */
if (c->stream->feed) {
ffm_set_write_index(c->fmt_in,
c->stream->feed->feed_write_index,
c->stream->feed->feed_size);
}
if (c->stream->max_time &&
c->stream->max_time + c->start_time - cur_time < 0) {
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
if (1 || c->is_packetized) {
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committed
if (compute_send_delay(c) > 0) {
c->state = HTTPSTATE_WAIT;
return 1; /* state changed */
}
}
redo:
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committed
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
/* if coming from feed, it means we reached the end of the
ffm file, so must wait for more data */
c->state = HTTPSTATE_WAIT_FEED;
return 1; /* state changed */
} else {
if (c->stream->loop) {
av_close_input_file(c->fmt_in);
c->fmt_in = NULL;
if (open_input_stream(c, "") < 0)
goto no_loop;
goto redo;
} else {
no_loop:
/* must send trailer now because eof or error */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
}
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}
} else {
/* update first pts if needed */
if (c->first_pts == AV_NOPTS_VALUE)
c->first_pts = pkt.pts;
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
if (c->switch_pending) {
c->switch_pending = 0;
for(i=0;i<c->stream->nb_streams;i++) {
if (c->switch_feed_streams[i] == pkt.stream_index) {
if (pkt.flags & PKT_FLAG_KEY) {
do_switch_stream(c, i);
}
}
if (c->switch_feed_streams[i] >= 0) {
c->switch_pending = 1;
}
}
}
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committed
if (c->feed_streams[i] == pkt.stream_index) {
pkt.stream_index = i;
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c->got_key_frame |= 1 << i;
}
/* See if we have all the key frames, then
* we start to send. This logic is not quite
* right, but it works for the case of a
* single video stream with one or more
* audio streams (for which every frame is
* typically a key frame).
*/
if (!c->stream->send_on_key ||
((c->got_key_frame + 1) >> c->stream->nb_streams)) {
goto send_it;
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committed
} else {
AVCodecContext *codec;
send_it:
/* specific handling for RTP: we use several
output stream (one for each RTP
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