#include <sys/poll.h>
#include <errno.h>
#include <sys/time.h>
+#undef time //needed because HAVE_AV_CONFIG_H is defined on top
#include <time.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
-#include <ctype.h>
#include <signal.h>
#ifdef CONFIG_HAVE_DLFCN
#include <dlfcn.h>
HTTPSTATE_SEND_DATA_TRAILER,
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,
+ RTSPSTATE_SEND_PACKET,
};
const char *http_state[] = {
"SEND_DATA_TRAILER",
"RECEIVE_DATA",
"WAIT_FEED",
- "WAIT",
- "WAIT_SHORT",
"READY",
"RTSP_WAIT_REQUEST",
"RTSP_SEND_REPLY",
+ "RTSP_SEND_PACKET",
};
#define IOBUFFER_INIT_SIZE 8192
#define SYNC_TIMEOUT (10 * 1000)
typedef struct {
- INT64 count1, count2;
+ int64_t count1, count2;
long time1, time2;
} DataRateData;
struct sockaddr_in from_addr; /* origin */
struct pollfd *poll_entry; /* used when polling */
long timeout;
- UINT8 *buffer_ptr, *buffer_end;
+ uint8_t *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;
+ int64_t data_count;
/* feed input */
int feed_fd;
/* input format handling */
AVFormatContext *fmt_in;
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 */
+ int64_t first_pts; /* initial pts value */
+ int64_t cur_pts; /* current pts value from the stream in us */
+ int64_t cur_frame_duration; /* duration of the current frame in us */
+ int cur_frame_bytes; /* output frame size, needed to compute
+ the time at which we send each
+ packet */
+ int pts_stream_index; /* stream we choose as clock reference */
+ int64_t cur_clock; /* current clock reference value in us */
/* output format handling */
struct FFStream *stream;
/* -1 is invalid stream */
char method[16];
char url[128];
int buffer_size;
- UINT8 *buffer;
+ uint8_t *buffer;
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 */
+ uint8_t *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];
+
+ /* RTP/UDP specific */
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) */
+
+ /* RTP/TCP specific */
+ struct HTTPContext *rtsp_c;
+ uint8_t *packet_buffer, *packet_buffer_ptr, *packet_buffer_end;
} HTTPContext;
+static AVFrame dummy_frame;
+
/* each generated stream is described here */
enum StreamType {
STREAM_TYPE_LIVE,
typedef struct IPAddressACL {
struct IPAddressACL *next;
enum IPAddressAction action;
+ /* These are in host order */
struct in_addr first;
struct in_addr last;
} IPAddressACL;
char filename[1024]; /* stream filename */
struct FFStream *feed; /* feed we are using (can be null if
coming from file) */
+ AVFormatParameters *ap_in; /* input parameters */
+ AVInputFormat *ifmt; /* if non NULL, force input format */
AVOutputFormat *fmt;
IPAddressACL *acl;
int nb_streams;
/* feed specific */
int feed_opened; /* true if someone is writing to the feed */
int is_feed; /* true if it is a feed */
+ int readonly; /* True if writing is prohibited to the file */
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 */
+ int64_t bytes_served;
+ int64_t feed_max_size; /* maximum storage size */
+ int64_t feed_write_index; /* current write position in feed (it wraps round) */
+ int64_t feed_size; /* current size of feed */
struct FFStream *next_feed;
} FFStream;
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);
-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_options(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);
/* SDP handling */
-static int prepare_sdp_description(FFStream *stream, UINT8 **pbuffer,
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
struct in_addr my_ip);
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
- FFStream *stream, const char *session_id);
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol);
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr);
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c);
static const char *my_program_name;
static const char *my_program_dir;
static FILE *logfile = NULL;
-static void http_log(char *fmt, ...)
+static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
c->protocol, (c->http_error ? c->http_error : 200), c->data_count);
}
-static void update_datarate(DataRateData *drd, INT64 count)
+static void update_datarate(DataRateData *drd, int64_t count)
{
if (!drd->time1 && !drd->count1) {
drd->time1 = drd->time2 = cur_time;
}
/* In bytes per second */
-static int compute_datarate(DataRateData *drd, INT64 count)
+static int compute_datarate(DataRateData *drd, int64_t 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)
+static int get_longterm_datarate(DataRateData *drd, int64_t count)
{
/* You get the first 3 seconds flat out */
if (cur_time - drd->time1 < 3000)
/* This is needed to make relative pathnames work */
chdir(my_program_dir);
+ signal(SIGPIPE, SIG_DFL);
+
execvp(pathname, feed->child_argv);
_exit(1);
setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
- perror ("bind");
+ char bindmsg[32];
+ snprintf(bindmsg, sizeof(bindmsg), "bind(port %d)", ntohs(my_addr->sin_port));
+ perror (bindmsg);
close(server_fd);
return -1;
}
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);
+ rtp_c = rtp_new_connection(&dest_addr, stream, session_id,
+ RTSP_PROTOCOL_RTP_UDP_MULTICAST);
if (!rtp_c) {
continue;
}
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) {
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, NULL) < 0) {
fprintf(stderr, "Could not open output stream '%s/streamid=%d'\n",
stream->filename, stream_index);
exit(1);
first_http_ctx = NULL;
nb_connections = 0;
- first_http_ctx = NULL;
start_multicast();
switch(c->state) {
case HTTPSTATE_SEND_HEADER:
case RTSPSTATE_SEND_REPLY:
+ case RTSPSTATE_SEND_PACKET:
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLOUT;
poll_entry->events = POLLOUT;
poll_entry++;
} else {
- /* not strictly correct, but currently cannot add
- more than one fd in poll entry */
- delay = 0;
+ /* when ffserver is doing the timing, we work by
+ looking at which packet need to be sent every
+ 10 ms */
+ delay1 = 10; /* one tick wait XXX: 10 ms assumed */
+ if (delay1 < delay)
+ delay = delay1;
}
break;
case HTTPSTATE_WAIT_REQUEST:
poll_entry->events = POLLIN;/* Maybe this will work */
poll_entry++;
break;
- 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;
second to handle timeouts */
do {
ret = poll(poll_table, poll_entry - poll_table, delay);
- } while (ret == -1);
+ if (ret < 0 && errno != EAGAIN && errno != EINTR)
+ return -1;
+ } while (ret <= 0);
cur_time = gettime_ms();
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 = av_malloc(c->buffer_size);
if (!c->buffer)
goto fail;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
nb_connections++;
start_wait_request(c, is_rtsp);
}
}
+ /* remove references, if any (XXX: do it faster) */
+ for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
+ if (c1->rtsp_c == c)
+ c1->rtsp_c = NULL;
+ }
+
/* remove connection associated resources */
if (c->fd >= 0)
close(c->fd);
url_close(h);
}
}
+
+ ctx = &c->fmt_ctx;
+
+ if (!c->last_packet_sent) {
+ if (ctx->oformat) {
+ /* prepare header */
+ if (url_open_dyn_buf(&ctx->pb) >= 0) {
+ av_write_trailer(ctx);
+ url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ }
+ }
+ }
+
+ for(i=0; i<ctx->nb_streams; i++)
+ av_free(ctx->streams[i]) ;
if (c->stream)
current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
+ av_freep(&c->packet_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
if (!(c->poll_entry->revents & POLLIN))
return 0;
/* read the data */
- len = read(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ read_loop:
+ len = read(c->fd, c->buffer_ptr, 1);
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;
+ /* search for end of request. */
+ uint8_t *ptr;
c->buffer_ptr += len;
ptr = c->buffer_ptr;
if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
- }
+ } else goto read_loop;
}
break;
/* nothing to do, we'll be waken up by incoming feed packets */
break;
- 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);
}
}
break;
+ case RTSPSTATE_SEND_PACKET:
+ if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = write(c->fd, c->packet_buffer_ptr,
+ c->packet_buffer_end - c->packet_buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ av_freep(&c->packet_buffer);
+ return -1;
+ }
+ } else {
+ c->packet_buffer_ptr += len;
+ if (c->packet_buffer_ptr >= c->packet_buffer_end) {
+ /* all the buffer was sent : wait for a new request */
+ av_freep(&c->packet_buffer);
+ c->state = RTSPSTATE_WAIT_REQUEST;
+ }
+ }
+ break;
case HTTPSTATE_READY:
/* nothing to do */
break;
FFStream *req = c->stream;
int action_required = 0;
+ /* Not much we can do for a feed */
+ if (!req->feed)
+ return 0;
+
for (i = 0; i < req->nb_streams; i++) {
AVCodecContext *codec = &req->streams[i]->codec;
enum IPAddressAction last_action = IP_DENY;
IPAddressACL *acl;
struct in_addr *src = &c->from_addr.sin_addr;
+ unsigned long src_addr = ntohl(src->s_addr);
for (acl = stream->acl; acl; acl = acl->next) {
- if (src->s_addr >= acl->first.s_addr && src->s_addr <= acl->last.s_addr) {
+ if (src_addr >= acl->first.s_addr && src_addr <= acl->last.s_addr) {
return (acl->action == IP_ALLOW) ? 1 : 0;
}
last_action = acl->action;
if (match_ext(filename, "asx")) {
redir_type = REDIR_ASX;
filename[strlen(filename)-1] = 'f';
- } else if (match_ext(filename, ".asf") &&
+ } else if (match_ext(filename, "asf") &&
(!useragent || strncasecmp(useragent, "NSPlayer", 8) != 0)) {
/* if this isn't WMP or lookalike, return the redirector file */
redir_type = REDIR_ASF;
stream = stream->next;
}
if (stream == NULL) {
- sprintf(msg, "File '%s' not found", url);
+ snprintf(msg, sizeof(msg), "File '%s' not found", url);
goto send_error;
}
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 += s
printf(q, "You should be <a href=\"%s\">redirected</a>.\r\n", stream->feed_filename);
- q += sprintf(q, "</body></html>\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 301 Moved\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Location: %s\r\n", stream->feed_filename);
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: text/html\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<html><head><title>Moved</title></head><body>\r\n");
+ q += s
nprintf(q, q - (char *) c->buffer + c->buffer_size, "You should be <a href=\"%s\">redirected</a>.\r\n", stream->feed_filename);
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "</body></html>\r\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
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 += s
printf(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",
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 Server too busy\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: text/html\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<html><head><title>Too busy</title></head><body>\r\n");
+ q += s
nprintf(q, q - (char *) c->buffer + c->buffer_size, "The server is too busy to serve your request at this time.<p>\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "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");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "</body></html>\r\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
q = c->buffer;
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 += s
printf(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",
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 ASX Follows\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: video/x-ms-asf\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += s
nprintf(q, q - (char *) c->buffer + c->buffer_size, "<ASX Version=\"3\">\r\n");
+ //q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<!-- Autogenerated by ffserver -->\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<ENTRY><REF HREF=\"http://%s/%s%s\"/></ENTRY>\r\n",
hostbuf, filename, info);
- q += sprintf(q, "</ASX>\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "</ASX>\r\n");
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",
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 RAM Follows\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: audio/x-pn-realaudio\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "# Autogenerated by ffserver\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "http://%s/%s%s\r\n",
hostbuf, filename, info);
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",
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 ASF Redirect follows\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: video/x-ms-asf\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "[Reference]\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Ref1=http://%s/%s%s\r\n",
hostbuf, filename, info);
break;
case REDIR_RTSP:
p = strrchr(hostname, ':');
if (p)
*p = '\0';
- q += sprintf(q, "HTTP/1.0 200 RTSP Redirect follows\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "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",
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: application/x-rtsp\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "rtsp://%s:%d/%s\r\n",
hostname, ntohs(my_rtsp_addr.sin_port),
filename);
}
break;
case REDIR_SDP:
{
- UINT8 *sdp_data;
+ uint8_t *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");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 OK\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: application/sdp\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
}
}
- sprintf(msg, "ASX/RAM file not handled");
+ snprintf(msg, sizeof(msg), "ASX/RAM file not handled");
goto send_error;
}
if (eol) {
if (eol[-1] == '\r')
eol--;
- http_log("%.*s\n", eol - logline, logline);
+ http_log("%.*s\n", (int) (eol - logline), logline);
c->suppress_log = 1;
}
}
}
}
- sprintf(msg, "POST command not handled");
+ snprintf(msg, sizeof(msg), "POST command not handled");
+ c->stream = 0;
goto send_error;
}
if (http_start_receive_data(c) < 0) {
- sprintf(msg, "could not open feed");
+ snprintf(msg, sizeof(msg), "could not open feed");
goto send_error;
}
c->http_error = 0;
/* open input stream */
if (open_input_stream(c, info) < 0) {
- sprintf(msg, "Input stream corresponding to '%s' not found", url);
+ snprintf(msg, sizeof(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");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "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");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Pragma: no-cache\r\n");
/* for asf, we need extra headers */
if (!strcmp(c->stream->fmt->name,"asf_stream")) {
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);
- mime_type = "application/octet-stream";
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "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);
}
- q += sprintf(q, "Content-Type: %s\r\n", mime_type);
- q += sprintf(q, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-Type: %s\r\n", mime_type);
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
/* prepare output buffer */
c->http_error = 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");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 404 Not Found\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-type: %s\r\n", "text/html");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<HTML>\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<HEAD><TITLE>404 Not Found</TITLE></HEAD>\n");
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<BODY>%s</BODY>\n", msg);
+ q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "</HTML>\n");
/* prepare output buffer */
c->buffer_ptr = c->buffer;
return 0;
}
-static void fmt_bytecount(ByteIOContext *pb, INT64 count)
+static void fmt_bytecount(ByteIOContext *pb, int64_t count)
{
static const char *suffix = " kMGTP";
const char *s;
video_codec_name = codec->name;
}
break;
+ case CODEC_TYPE_DATA:
+ video_bit_rate += st->codec.bit_rate;
+ break;
default:
av_abort();
}
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";
+ const char *type = "unknown";
char parameters[64];
parameters[0] = 0;
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);
+ snprintf(parameters, sizeof(parameters), "%dx%d, q=%d-%d, fps=%d", st->codec.width, st->codec.height,
+ st->codec.qmin, st->codec.qmax, st->codec.time_base.den / st->codec.time_base.num);
break;
default:
av_abort();
char input_filename[1024];
AVFormatContext *s;
int buf_size, i;
- INT64 stream_pos;
+ int64_t stream_pos;
/* find file name */
if (c->stream->feed) {
stream_pos = parse_date(buf, 0);
} 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() - prebuffer * (int64_t)1000000;
} else {
- stream_pos = av_gettime() - c->stream->prebuffer * (INT64)1000;
+ stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
}
} else {
strcpy(input_filename, c->stream->feed_filename);
#endif
/* open stream */
- if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 0) {
+ if (av_open_input_file(&s, input_filename, c->stream->ifmt,
+ buf_size, c->stream->ap_in) < 0) {
http_log("%s not found", input_filename);
return -1;
}
}
}
+#if 1
if (c->fmt_in->iformat->read_seek) {
- c->fmt_in->iformat->read_seek(c->fmt_in, stream_pos);
+ c->fmt_in->iformat->read_seek(c->fmt_in, 0, stream_pos, 0);
}
+#endif
/* set the start time (needed for maxtime and RTP packet timing) */
c->start_time = cur_time;
c->first_pts = AV_NOPTS_VALUE;
return 0;
}
-/* currently desactivated because the new PTS handling is not
- satisfactory yet */
-//#define AV_READ_FRAME
-#ifdef AV_READ_FRAME
-
-/* 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)
+/* return the server clock (in us) */
+static int64_t get_server_clock(HTTPContext *c)
{
- AVStream *st;
- int len, ret, old_nb_streams, i;
-
- /* 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;
- if (st->codec.coded_frame->key_frame)
- pkt->flags |= PKT_FLAG_KEY;
- return 0;
- }
- }
- } else {
- /* 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;
- }
- }
- }
+ /* compute current pts value from system time */
+ return (int64_t)(cur_time - c->start_time) * 1000LL;
}
-static int compute_send_delay(HTTPContext *c)
+/* return the estimated time at which the current packet must be sent
+ (in us) */
+static int64_t get_packet_send_clock(HTTPContext *c)
{
- INT64 cur_pts, delta_pts, next_pts;
- int delay1;
+ int bytes_left, bytes_sent, frame_bytes;
- /* 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;
+ frame_bytes = c->cur_frame_bytes;
+ if (frame_bytes <= 0) {
+ return c->cur_pts;
} else {
- delay1 = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
+ bytes_left = c->buffer_end - c->buffer_ptr;
+ bytes_sent = frame_bytes - bytes_left;
+ return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
}
- 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;
- }
- return 0;
-}
-#endif
-
static int http_prepare_data(HTTPContext *c)
{
int i, len, ret;
AVFormatContext *ctx;
+ av_freep(&c->pb_buffer);
switch(c->state) {
case HTTPSTATE_SEND_DATA_HEADER:
memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
c->fmt_ctx.nb_streams = c->stream->nb_streams;
for(i=0;i<c->fmt_ctx.nb_streams;i++) {
AVStream *st;
+ AVStream *src;
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));
+ src = c->stream->streams[i];
else
- memcpy(st, c->stream->feed->streams[c->stream->feed_streams[i]],
- sizeof(AVStream));
+ src = c->stream->feed->streams[c->stream->feed_streams[i]];
+
+ *st = *src;
+ st->priv_data = 0;
st->codec.frame_number = 0; /* XXX: should be done in
AVStream, not in codec */
+ /* I'm pretty sure that this is not correct...
+ * However, without it, we crash
+ */
+ st->codec.coded_frame = &dummy_frame;
}
c->got_key_frame = 0;
}
c->fmt_ctx.pb.is_streamed = 1;
+ av_set_parameters(&c->fmt_ctx, NULL);
av_write_header(&c->fmt_ctx);
len = url_close_dyn_buf(&c->fmt_ctx.pb, &c->pb_buffer);
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
} else {
- if (1 || c->is_packetized) {
- if (compute_send_delay(c) > 0) {
- c->state = HTTPSTATE_WAIT;
- return 1; /* state changed */
- }
- }
redo:
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
}
} else {
/* update first pts if needed */
- if (c->first_pts == AV_NOPTS_VALUE)
- c->first_pts = pkt.pts;
-
+ if (c->first_pts == AV_NOPTS_VALUE) {
+ c->first_pts = av_rescale_q(pkt.dts, c->fmt_in->streams[pkt.stream_index]->time_base, AV_TIME_BASE_Q);
+ c->start_time = cur_time;
+ }
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
output stream (one for each RTP
connection). XXX: need more abstract handling */
if (c->is_packetized) {
+ AVStream *st;
+ /* compute send time and duration */
+ st = c->fmt_in->streams[pkt.stream_index];
+ c->cur_pts = av_rescale_q(pkt.dts, st->time_base, AV_TIME_BASE_Q);
+ if (st->start_time != AV_NOPTS_VALUE)
+ c->cur_pts -= av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q);
+ c->cur_frame_duration = av_rescale_q(pkt.duration, st->time_base, AV_TIME_BASE_Q);
+#if 0
+ printf("index=%d pts=%0.3f duration=%0.6f\n",
+ pkt.stream_index,
+ (double)c->cur_pts /
+ AV_TIME_BASE,
+ (double)c->cur_frame_duration /
+ AV_TIME_BASE);
+#endif
+ /* find RTP context */
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
+ if(!ctx) {
+ av_free_packet(&pkt);
+ break;
+ }
codec = &ctx->streams[0]->codec;
/* only one stream per RTP connection */
pkt.stream_index = 0;
}
codec->coded_frame->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
-
-#ifdef PJSG
- if (codec->codec_type == CODEC_TYPE_AUDIO) {
- codec->frame_size = (codec->sample_rate * pkt.duration + 500000) / 1000000;
- /* printf("Calculated size %d, from sr %d, duration %d\n", codec->frame_size, codec->sample_rate, pkt.duration); */
- }
-#endif
-
if (c->is_packetized) {
- ret = url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]));
- c->packet_byte_count = 0;
- c->packet_start_time_us = av_gettime();
+ int max_packet_size;
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ else
+ max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
+ ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
} else {
ret = url_open_dyn_buf(&ctx->pb);
}
/* XXX: potential leak */
return -1;
}
- if (av_write_frame(ctx, pkt.stream_index, pkt.data, pkt.size)) {
+ if (av_write_frame(ctx, &pkt)) {
c->state = HTTPSTATE_SEND_DATA_TRAILER;
}
len = url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ c->cur_frame_bytes = len;
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
codec->frame_number++;
+ if (len == 0)
+ goto redo;
}
-#ifndef AV_READ_FRAME
av_free_packet(&pkt);
-#endif
}
}
}
#define SHORT_TERM_BANDWIDTH 8000000
/* should convert the format at the same time */
+/* send data starting at c->buffer_ptr to the output connection
+ (either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
- int len, ret, dt;
-
- while (c->buffer_ptr >= c->buffer_end) {
- av_freep(&c->pb_buffer);
- ret = http_prepare_data(c);
- if (ret < 0)
- return -1;
- else if (ret == 0) {
- continue;
- } else {
- /* state change requested */
- return 0;
- }
- }
+ int len, ret;
- if (c->buffer_ptr < c->buffer_end) {
- if (c->is_packetized) {
- /* RTP/UDP data output */
- len = c->buffer_end - c->buffer_ptr;
- if (len < 4) {
- /* fail safe - should never happen */
- fail1:
- c->buffer_ptr = c->buffer_end;
- return 0;
- }
- len = (c->buffer_ptr[0] << 24) |
- (c->buffer_ptr[1] << 16) |
- (c->buffer_ptr[2] << 8) |
- (c->buffer_ptr[3]);
- if (len > (c->buffer_end - c->buffer_ptr))
- goto fail1;
-
- /* short term bandwidth limitation */
- dt = av_gettime() - c->packet_start_time_us;
- if (dt < 1)
- dt = 1;
-
- if ((c->packet_byte_count + len) * (INT64)1000000 >=
- (SHORT_TERM_BANDWIDTH / 8) * (INT64)dt) {
- /* bandwidth overflow : wait at most one tick and retry */
- c->state = HTTPSTATE_WAIT_SHORT;
- return 0;
+ for(;;) {
+ if (c->buffer_ptr >= c->buffer_end) {
+ ret = http_prepare_data(c);
+ if (ret < 0)
+ return -1;
+ else if (ret != 0) {
+ /* state change requested */
+ break;
}
-
- c->buffer_ptr += 4;
- url_write(c->rtp_handles[c->packet_stream_index],
- c->buffer_ptr, len);
- c->buffer_ptr += len;
- c->packet_byte_count += len;
} else {
- /* TCP data output */
- len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
- if (len < 0) {
- if (errno != EAGAIN && errno != EINTR) {
- /* error : close connection */
- return -1;
- } else {
+ if (c->is_packetized) {
+ /* RTP data output */
+ len = c->buffer_end - c->buffer_ptr;
+ if (len < 4) {
+ /* fail safe - should never happen */
+ fail1:
+ c->buffer_ptr = c->buffer_end;
return 0;
}
+ len = (c->buffer_ptr[0] << 24) |
+ (c->buffer_ptr[1] << 16) |
+ (c->buffer_ptr[2] << 8) |
+ (c->buffer_ptr[3]);
+ if (len > (c->buffer_end - c->buffer_ptr))
+ goto fail1;
+ if ((get_packet_send_clock(c) - get_server_clock(c)) > 0) {
+ /* nothing to send yet: we can wait */
+ return 0;
+ }
+
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+
+ if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
+ /* RTP packets are sent inside the RTSP TCP connection */
+ ByteIOContext pb1, *pb = &pb1;
+ int interleaved_index, size;
+ uint8_t header[4];
+ HTTPContext *rtsp_c;
+
+ rtsp_c = c->rtsp_c;
+ /* if no RTSP connection left, error */
+ if (!rtsp_c)
+ return -1;
+ /* if already sending something, then wait. */
+ if (rtsp_c->state != RTSPSTATE_WAIT_REQUEST) {
+ break;
+ }
+ if (url_open_dyn_buf(pb) < 0)
+ goto fail1;
+ interleaved_index = c->packet_stream_index * 2;
+ /* RTCP packets are sent at odd indexes */
+ if (c->buffer_ptr[1] == 200)
+ interleaved_index++;
+ /* write RTSP TCP header */
+ header[0] = '$';
+ header[1] = interleaved_index;
+ header[2] = len >> 8;
+ header[3] = len;
+ put_buffer(pb, header, 4);
+ /* write RTP packet data */
+ c->buffer_ptr += 4;
+ put_buffer(pb, c->buffer_ptr, len);
+ size = url_close_dyn_buf(pb, &c->packet_buffer);
+ /* prepare asynchronous TCP sending */
+ rtsp_c->packet_buffer_ptr = c->packet_buffer;
+ rtsp_c->packet_buffer_end = c->packet_buffer + size;
+ c->buffer_ptr += len;
+
+ /* send everything we can NOW */
+ len = write(rtsp_c->fd, rtsp_c->packet_buffer_ptr,
+ rtsp_c->packet_buffer_end - rtsp_c->packet_buffer_ptr);
+ if (len > 0) {
+ rtsp_c->packet_buffer_ptr += len;
+ }
+ if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
+ /* if we could not send all the data, we will
+ send it later, so a new state is needed to
+ "lock" the RTSP TCP connection */
+ rtsp_c->state = RTSPSTATE_SEND_PACKET;
+ break;
+ } else {
+ /* all data has been sent */
+ av_freep(&c->packet_buffer);
+ }
+ } else {
+ /* send RTP packet directly in UDP */
+ c->buffer_ptr += 4;
+ url_write(c->rtp_handles[c->packet_stream_index],
+ c->buffer_ptr, len);
+ c->buffer_ptr += len;
+ /* here we continue as we can send several packets per 10 ms slot */
+ }
} else {
- c->buffer_ptr += len;
+ /* TCP data output */
+ len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ return -1;
+ } else {
+ return 0;
+ }
+ } else {
+ c->buffer_ptr += len;
+ }
+ c->data_count += len;
+ update_datarate(&c->datarate, c->data_count);
+ if (c->stream)
+ c->stream->bytes_served += len;
+ break;
}
}
- c->data_count += len;
- update_datarate(&c->datarate, c->data_count);
- if (c->stream)
- c->stream->bytes_served += len;
- }
+ } /* for(;;) */
return 0;
}
if (c->stream->feed_opened)
return -1;
+ /* Don't permit writing to this one */
+ if (c->stream->readonly)
+ return -1;
+
/* open feed */
fd = open(c->stream->feed_filename, O_RDWR);
if (fd < 0)
}
}
+ if (c->buffer_ptr - c->buffer >= 2 && c->data_count > FFM_PACKET_SIZE) {
+ if (c->buffer[0] != 'f' ||
+ c->buffer[1] != 'm') {
+ http_log("Feed stream has become desynchronized -- disconnecting\n");
+ goto fail;
+ }
+ }
+
if (c->buffer_ptr >= c->buffer_end) {
FFStream *feed = c->stream;
/* a packet has been received : write it in the store, except
if (!fmt_in)
goto fail;
- s.priv_data = av_mallocz(fmt_in->priv_data_size);
- if (!s.priv_data)
- goto fail;
+ if (fmt_in->priv_data_size > 0) {
+ s.priv_data = av_mallocz(fmt_in->priv_data_size);
+ if (!s.priv_data)
+ goto fail;
+ } else
+ s.priv_data = NULL;
if (fmt_in->read_header(&s, 0) < 0) {
av_freep(&s.priv_data);
if (!strcmp(cmd, "DESCRIBE")) {
rtsp_cmd_describe(c, url);
+ } else if (!strcmp(cmd, "OPTIONS")) {
+ rtsp_cmd_options(c, url);
} else if (!strcmp(cmd, "SETUP")) {
rtsp_cmd_setup(c, url, header);
} else if (!strcmp(cmd, "PLAY")) {
/* XXX: move that to rtsp.c, but would need to replace FFStream by
AVFormatContext */
-static int prepare_sdp_description(FFStream *stream, UINT8 **pbuffer,
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
struct in_addr my_ip)
{
ByteIOContext pb1, *pb = &pb1;
url_fprintf(pb, "c=IN IP4 %s\n", inet_ntoa(stream->multicast_ip));
}
/* for each stream, we output the necessary info */
- private_payload_type = 96;
+ private_payload_type = RTP_PT_PRIVATE;
for(i = 0; i < stream->nb_streams; i++) {
st = stream->streams[i];
- switch(st->codec.codec_type) {
- case CODEC_TYPE_AUDIO:
- mediatype = "audio";
- break;
- case CODEC_TYPE_VIDEO:
+ if (st->codec.codec_id == CODEC_ID_MPEG2TS) {
mediatype = "video";
- break;
- default:
- mediatype = "application";
- break;
+ } else {
+ switch(st->codec.codec_type) {
+ case CODEC_TYPE_AUDIO:
+ mediatype = "audio";
+ break;
+ case CODEC_TYPE_VIDEO:
+ mediatype = "video";
+ break;
+ default:
+ mediatype = "application";
+ break;
+ }
}
/* NOTE: the port indication is not correct in case of
unicast. It is not an issue because RTSP gives it */
}
url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
mediatype, port, payload_type);
- if (payload_type >= 96) {
+ if (payload_type >= RTP_PT_PRIVATE) {
/* for private payload type, we need to give more info */
switch(st->codec.codec_id) {
case CODEC_ID_MPEG4:
/* we must also add the mpeg4 header */
data = st->codec.extradata;
if (data) {
- url_fprintf(pb, "a=fmtp:%d config=");
+ url_fprintf(pb, "a=fmtp:%d config=", payload_type);
for(j=0;j<st->codec.extradata_size;j++) {
url_fprintf(pb, "%02x", data[j]);
}
return -1;
}
+static void rtsp_cmd_options(HTTPContext *c, const char *url)
+{
+// rtsp_reply_header(c, RTSP_STATUS_OK);
+ url_fprintf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
+ url_fprintf(c->pb, "CSeq: %d\r\n", c->seq);
+ url_fprintf(c->pb, "Public: %s\r\n", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE");
+ url_fprintf(c->pb, "\r\n");
+}
+
static void rtsp_cmd_describe(HTTPContext *c, const char *url)
{
FFStream *stream;
char path1[1024];
const char *path;
- UINT8 *content;
+ uint8_t *content;
int content_length, len;
struct sockaddr_in my_addr;
/* find which url is asked */
- url_split(NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
path = path1;
if (*path == '/')
path++;
/* get the host IP */
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
-
content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
if (content_length < 0) {
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return NULL;
}
-RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
+static RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
{
RTSPTransportField *th;
int i;
RTSPActionServerSetup setup;
/* find which url is asked */
- url_split(NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
path = path1;
if (*path == '/')
path++;
/* find rtp session, and create it if none found */
rtp_c = find_rtp_session(h->session_id);
if (!rtp_c) {
- rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
+ /* always prefer UDP */
+ th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
+ if (!th) {
+ th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
+ if (!th) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ }
+
+ rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id,
+ th->protocol);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
}
-
- /* always prefer UDP */
- th = find_transport(h, RTSP_PROTOCOL_RTP_UDP);
- if (!th) {
- th = find_transport(h, RTSP_PROTOCOL_RTP_TCP);
- if (!th) {
- rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
- return;
- }
- }
- rtp_c->rtp_protocol = th->protocol;
}
/* test if stream is OK (test needed because several SETUP needs
}
/* setup stream */
- if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr, c) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
HTTPContext *rtp_c;
char path1[1024];
const char *path;
+ char buf[1024];
+ int s;
rtp_c = find_rtp_session(session_id);
if (!rtp_c)
return NULL;
/* find which url is asked */
- url_split(NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
+ url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
path = path1;
if (*path == '/')
path++;
- if (strcmp(path, rtp_c->stream->filename) != 0)
- return NULL;
- return rtp_c;
+ if(!strcmp(path, rtp_c->stream->filename)) return rtp_c;
+ for(s=0; s<rtp_c->stream->nb_streams; ++s) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ rtp_c->stream->filename, s);
+ if(!strncmp(path, buf, sizeof(buf))) {
+ // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
+ return rtp_c;
+ }
+ }
+ return NULL;
}
static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
return;
}
+#if 0
+ /* XXX: seek in stream */
+ if (h->range_start != AV_NOPTS_VALUE) {
+ printf("range_start=%0.3f\n", (double)h->range_start / AV_TIME_BASE);
+ av_seek_frame(rtp_c->fmt_in, -1, h->range_start);
+ }
+#endif
+
rtp_c->state = HTTPSTATE_SEND_DATA;
/* now everything is OK, so we can send the connection parameters */
}
rtp_c->state = HTTPSTATE_READY;
-
+ rtp_c->first_pts = AV_NOPTS_VALUE;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
/* RTP handling */
static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
- FFStream *stream, const char *session_id)
+ FFStream *stream, const char *session_id,
+ enum RTSPProtocol rtp_protocol)
{
HTTPContext *c = NULL;
-
+ const char *proto_str;
+
/* XXX: should output a warning page when coming
close to the connection limit */
if (nb_connections >= nb_max_connections)
pstrcpy(c->session_id, sizeof(c->session_id), session_id);
c->state = HTTPSTATE_READY;
c->is_packetized = 1;
+ c->rtp_protocol = rtp_protocol;
+
/* protocol is shown in statistics */
- pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ proto_str = "MCAST";
+ break;
+ case RTSP_PROTOCOL_RTP_UDP:
+ proto_str = "UDP";
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ proto_str = "TCP";
+ break;
+ default:
+ proto_str = "???";
+ break;
+ }
+ pstrcpy(c->protocol, sizeof(c->protocol), "RTP/");
+ pstrcat(c->protocol, sizeof(c->protocol), proto_str);
current_bandwidth += stream->bandwidth;
}
/* add a new RTP stream in an RTP connection (used in RTSP SETUP
- command). if dest_addr is NULL, then TCP tunneling in RTSP is
+ command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
used. */
static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr)
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
{
AVFormatContext *ctx;
AVStream *st;
char *ipaddr;
URLContext *h;
- UINT8 *dummy_buf;
+ uint8_t *dummy_buf;
char buf2[32];
+ int max_packet_size;
/* now we can open the relevant output stream */
- ctx = av_mallocz(sizeof(AVFormatContext));
+ ctx = av_alloc_format_context();
if (!ctx)
return -1;
ctx->oformat = &rtp_mux;
sizeof(AVStream));
}
- if (dest_addr) {
- /* build destination RTP address */
- ipaddr = inet_ntoa(dest_addr->sin_addr);
+ /* build destination RTP address */
+ ipaddr = inet_ntoa(dest_addr->sin_addr);
+
+ switch(c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP:
+ case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+ /* RTP/UDP case */
/* XXX: also pass as parameter to function ? */
if (c->stream->is_multicast) {
if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
goto fail;
c->rtp_handles[stream_index] = h;
- } else {
+ max_packet_size = url_get_max_packet_size(h);
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ /* RTP/TCP case */
+ c->rtsp_c = rtsp_c;
+ max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
+ break;
+ default:
goto fail;
}
- http_log("%s:%d - - [%s] \"RTPSTART %s/streamid=%d\"\n",
+ http_log("%s:%d - - [%s] \"PLAY %s/streamid=%d %s\"\n",
ipaddr, ntohs(dest_addr->sin_port),
ctime1(buf2),
- c->stream->filename, stream_index);
+ c->stream->filename, stream_index, c->protocol);
/* normally, no packets should be output here, but the packet size may be checked */
- if (url_open_dyn_packet_buf(&ctx->pb,
- url_get_max_packet_size(h)) < 0) {
+ if (url_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
/* XXX: close stream */
goto fail;
}
+ av_set_parameters(ctx, NULL);
if (av_write_header(ctx) < 0) {
fail:
if (h)
/********************************************************************/
/* ffserver initialization */
-AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
+static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
{
AVStream *fst;
return NULL;
fst->priv_data = av_mallocz(sizeof(FeedData));
memcpy(&fst->codec, codec, sizeof(AVCodecContext));
+ fst->codec.coded_frame = &dummy_frame;
+ fst->index = stream->nb_streams;
+ av_set_pts_info(fst, 33, 1, 90000);
stream->streams[stream->nb_streams++] = fst;
return fst;
}
/* return the stream number in the feed */
-int add_av_stream(FFStream *feed,
- AVStream *st)
+static int add_av_stream(FFStream *feed, AVStream *st)
{
AVStream *fst;
AVCodecContext *av, *av1;
case CODEC_TYPE_VIDEO:
if (av1->width == av->width &&
av1->height == av->height &&
- av1->frame_rate == av->frame_rate &&
+ av1->time_base.den == av->time_base.den &&
+ av1->time_base.num == av->time_base.num &&
av1->gop_size == av->gop_size)
goto found;
break;
return i;
}
-void remove_stream(FFStream *stream)
+static void remove_stream(FFStream *stream)
{
FFStream **ps;
ps = &first_stream;
}
/* specific mpeg4 handling : we extract the raw parameters */
-void extract_mpeg4_header(AVFormatContext *infile)
+static void extract_mpeg4_header(AVFormatContext *infile)
{
int mpeg4_count, i, size;
AVPacket pkt;
AVStream *st;
- const UINT8 *p;
+ const uint8_t *p;
mpeg4_count = 0;
for(i=0;i<infile->nb_streams;i++) {
st = infile->streams[i];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
- st->codec.extradata == NULL) {
+ st->codec.extradata_size == 0) {
mpeg4_count++;
}
}
if (!mpeg4_count)
return;
- printf("MPEG4 without extra data: trying to find header\n");
+ printf("MPEG4 without extra data: trying to find header in %s\n", infile->filename);
while (mpeg4_count > 0) {
if (av_read_packet(infile, &pkt) < 0)
break;
st = infile->streams[pkt.stream_index];
if (st->codec.codec_id == CODEC_ID_MPEG4 &&
- st->codec.extradata == NULL) {
+ st->codec.extradata_size == 0) {
+ av_freep(&st->codec.extradata);
/* fill extradata with the header */
/* XXX: we make hard suppositions here ! */
p = pkt.data;
}
/* compute the needed AVStream for each file */
-void build_file_streams(void)
+static void build_file_streams(void)
{
FFStream *stream, *stream_next;
AVFormatContext *infile;
/* the stream comes from a file */
/* try to open the file */
/* open stream */
+ stream->ap_in = av_mallocz(sizeof(AVFormatParameters));
+ if (stream->fmt == &rtp_mux) {
+ /* specific case : if transport stream output to RTP,
+ we use a raw transport stream reader */
+ stream->ap_in->mpeg2ts_raw = 1;
+ stream->ap_in->mpeg2ts_compute_pcr = 1;
+ }
+
if (av_open_input_file(&infile, stream->feed_filename,
- NULL, 0, NULL) < 0) {
+ stream->ifmt, 0, stream->ap_in) < 0) {
http_log("%s not found", stream->feed_filename);
/* remove stream (no need to spend more time on it) */
fail:
}
/* compute the needed AVStream for each feed */
-void build_feed_streams(void)
+static void build_feed_streams(void)
{
FFStream *stream, *feed;
int i;
if (sf->index != ss->index ||
sf->id != ss->id) {
- printf("Index & Id do not match for stream %d\n", i);
+ printf("Index & Id do not match for stream %d (%s)\n",
+ i, feed->feed_filename);
matches = 0;
} else {
AVCodecContext *ccf, *ccs;
printf("Codec bitrates do not match for stream %d\n", i);
matches = 0;
} else if (ccf->codec_type == CODEC_TYPE_VIDEO) {
- if (CHECK_CODEC(frame_rate) ||
+ if (CHECK_CODEC(time_base.den) ||
+ CHECK_CODEC(time_base.num) ||
CHECK_CODEC(width) ||
CHECK_CODEC(height)) {
printf("Codec width, height and framerate do not match for stream %d\n", i);
printf("Deleting feed file '%s' as it appears to be corrupt\n",
feed->feed_filename);
}
- if (!matches)
+ if (!matches) {
+ if (feed->readonly) {
+ printf("Unable to delete feed file '%s' as it is marked readonly\n",
+ feed->feed_filename);
+ exit(1);
+ }
unlink(feed->feed_filename);
+ }
}
if (!url_exist(feed->feed_filename)) {
AVFormatContext s1, *s = &s1;
+ if (feed->readonly) {
+ printf("Unable to create feed file '%s' as it is marked readonly\n",
+ feed->feed_filename);
+ exit(1);
+ }
+
/* only write the header of the ffm file */
if (url_fopen(&s->pb, feed->feed_filename, URL_WRONLY) < 0) {
fprintf(stderr, "Could not open output feed file '%s'\n",
st = feed->streams[i];
s->streams[i] = st;
}
+ av_set_parameters(s, NULL);
av_write_header(s);
/* XXX: need better api */
av_freep(&s->priv_data);
}
/* add a codec and set the default parameters */
-void add_codec(FFStream *stream, AVCodecContext *av)
+static void add_codec(FFStream *stream, AVCodecContext *av)
{
AVStream *st;
case CODEC_TYPE_VIDEO:
if (av->bit_rate == 0)
av->bit_rate = 64000;
- if (av->frame_rate == 0)
- av->frame_rate = 5 * FRAME_RATE_BASE;
+ if (av->time_base.num == 0){
+ av->time_base.den = 5;
+ av->time_base.num = 1;
+ }
if (av->width == 0 || av->height == 0) {
av->width = 160;
av->height = 128;
av->qcompress = 0.5;
av->qblur = 0.5;
+ if (!av->nsse_weight)
+ av->nsse_weight = 8;
+
+ av->frame_skip_cmp = FF_CMP_DCTMAX;
+ av->me_method = ME_EPZS;
+ av->rc_buffer_aggressivity = 1.0;
+
if (!av->rc_eq)
av->rc_eq = "tex^qComp";
if (!av->i_quant_factor)
av->b_quant_factor = 1.25;
if (!av->b_quant_offset)
av->b_quant_offset = 1.25;
- if (!av->rc_min_rate)
- av->rc_min_rate = av->bit_rate / 2;
if (!av->rc_max_rate)
av->rc_max_rate = av->bit_rate * 2;
+ if (av->rc_max_rate && !av->rc_buffer_size) {
+ av->rc_buffer_size = av->rc_max_rate;
+ }
+
+
break;
default:
av_abort();
memcpy(&st->codec, av, sizeof(AVCodecContext));
}
-int opt_audio_codec(const char *arg)
+static int opt_audio_codec(const char *arg)
{
AVCodec *p;
return p->id;
}
-int opt_video_codec(const char *arg)
+static int opt_video_codec(const char *arg)
{
AVCodec *p;
}
#endif
-int parse_ffconfig(const char *filename)
+static int parse_ffconfig(const char *filename)
{
FILE *f;
char line[1024];
if (!argbuf[0])
break;
- feed->child_argv[i] = av_malloc(strlen(argbuf + 1));
+ feed->child_argv[i] = av_malloc(strlen(argbuf) + 1);
strcpy(feed->child_argv[i], argbuf);
}
snprintf(feed->child_argv[i], 256, "http://127.0.0.1:%d/%s",
ntohs(my_http_addr.sin_port), feed->filename);
}
+ } else if (!strcasecmp(cmd, "ReadOnlyFile")) {
+ if (feed) {
+ get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
+ feed->readonly = 1;
+ } else if (stream) {
+ get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
+ }
} else if (!strcasecmp(cmd, "File")) {
if (feed) {
get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
fsize *= 1024 * 1024 * 1024;
break;
}
- feed->feed_max_size = (INT64)fsize;
+ feed->feed_max_size = (int64_t)fsize;
}
} else if (!strcasecmp(cmd, "</Feed>")) {
if (!feed) {
audio_id = stream->fmt->audio_codec;
video_id = stream->fmt->video_codec;
}
+ } else if (!strcasecmp(cmd, "InputFormat")) {
+ stream->ifmt = av_find_input_format(arg);
+ if (!stream->ifmt) {
+ fprintf(stderr, "%s:%d: Unknown input format: %s\n",
+ filename, line_num, arg);
+ }
} else if (!strcasecmp(cmd, "FaviconURL")) {
if (stream && stream->stream_type == STREAM_TYPE_STATUS) {
get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
errors++;
}
}
+ } else if (!strcasecmp(cmd, "Debug")) {
+ if (stream) {
+ get_arg(arg, sizeof(arg), &p);
+ video_enc.debug = strtol(arg,0,0);
+ }
+ } else if (!strcasecmp(cmd, "Strict")) {
+ if (stream) {
+ get_arg(arg, sizeof(arg), &p);
+ video_enc.strict_std_compliance = atoi(arg);
+ }
+ } else if (!strcasecmp(cmd, "VideoBufferSize")) {
+ if (stream) {
+ get_arg(arg, sizeof(arg), &p);
+ video_enc.rc_buffer_size = atoi(arg) * 8*1024;
+ }
} else if (!strcasecmp(cmd, "VideoBitRateTolerance")) {
if (stream) {
get_arg(arg, sizeof(arg), &p);
} else if (!strcasecmp(cmd, "VideoFrameRate")) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
- video_enc.frame_rate = (int)(strtod(arg, NULL) * FRAME_RATE_BASE);
+ video_enc.time_base.num= DEFAULT_FRAME_RATE_BASE;
+ video_enc.time_base.den = (int)(strtod(arg, NULL) * video_enc.time_base.num);
}
} else if (!strcasecmp(cmd, "VideoGopSize")) {
get_arg(arg, sizeof(arg), &p);
}
} else if (!strcasecmp(cmd, "VideoHighQuality")) {
if (stream) {
- video_enc.flags |= CODEC_FLAG_HQ;
+ video_enc.mb_decision = FF_MB_DECISION_BITS;
+ }
+ } else if (!strcasecmp(cmd, "Video4MotionVector")) {
+ if (stream) {
+ video_enc.mb_decision = FF_MB_DECISION_BITS; //FIXME remove
+ video_enc.flags |= CODEC_FLAG_4MV;
}
} else if (!strcasecmp(cmd, "VideoQDiff")) {
get_arg(arg, sizeof(arg), &p);
errors++;
} else {
/* Only take the first */
- acl.first = *(struct in_addr *) he->h_addr_list[0];
+ acl.first.s_addr = ntohl(((struct in_addr *) he->h_addr_list[0])->s_addr);
acl.last = acl.first;
}
errors++;
} else {
/* Only take the first */
- acl.last = *(struct in_addr *) he->h_addr_list[0];
+ acl.last.s_addr = ntohl(((struct in_addr *) he->h_addr_list[0])->s_addr);
}
}
#if 0
static void write_packet(FFCodec *ffenc,
- UINT8 *buf, int size)
+ uint8_t *buf, int size)
{
PacketHeader hdr;
AVCodecContext *enc = &ffenc->enc;
- UINT8 *wptr;
+ uint8_t *wptr;
mk_header(&hdr, enc, size);
wptr = http_fifo.wptr;
- fifo_write(&http_fifo, (UINT8 *)&hdr, sizeof(hdr), &wptr);
+ fifo_write(&http_fifo, (uint8_t *)&hdr, sizeof(hdr), &wptr);
fifo_write(&http_fifo, buf, size, &wptr);
/* atomic modification of wptr */
http_fifo.wptr = wptr;
}
#endif
-void help(void)
+static void show_banner(void)
+{
+ printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000-2003 Fabrice Bellard\n");
+}
+
+static void show_help(void)
{
- printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
- "usage: ffserver [-L] [-h] [-f configfile]\n"
+ show_banner();
+ printf("usage: ffserver [-L] [-h] [-f configfile]\n"
"Hyper fast multi format Audio/Video streaming server\n"
"\n"
- "-L : print the LICENCE\n"
+ "-L : print the LICENSE\n"
"-h : this help\n"
"-f configfile : use configfile instead of /etc/ffserver.conf\n"
);
}
-void licence(void)
+static void show_license(void)
{
+ show_banner();
printf(
- "ffserver version " FFMPEG_VERSION "\n"
- "Copyright (c) 2000, 2001, 2002 Fabrice Bellard\n"
"This library is free software; you can redistribute it and/or\n"
"modify it under the terms of the GNU Lesser General Public\n"
"License as published by the Free Software Foundation; either\n"
break;
switch(c) {
case 'L':
- licence();
+ show_license();
exit(1);
case '?':
case 'h':
- help();
+ show_help();
exit(1);
case 'n':
no_launch = 1;
projects / ffmpeg / blobdiff