* Multiple format streaming server
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* 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
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define HAVE_AV_CONFIG_H
-#include "common.h"
#include "avformat.h"
#include <stdarg.h>
#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
+#ifdef HAVE_DLFCN_H
#include <dlfcn.h>
#endif
+#include "version.h"
#include "ffserver.h"
/* maximum number of simultaneous HTTP connections */
HTTPSTATE_SEND_DATA_HEADER,
HTTPSTATE_SEND_DATA, /* sending TCP or UDP data */
HTTPSTATE_SEND_DATA_TRAILER,
- HTTPSTATE_RECEIVE_DATA,
+ 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
long timeout;
uint8_t *buffer_ptr, *buffer_end;
int http_error;
+ int post;
struct HTTPContext *next;
int got_key_frame; /* stream 0 => 1, stream 1 => 2, stream 2=> 4 */
int64_t data_count;
AVFormatContext *fmt_in;
long start_time; /* In milliseconds - this wraps fairly often */
int64_t first_pts; /* initial pts value */
- int pts_stream_index; /* stream we choose as clock reference */
+ 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 */
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_t *pb_buffer; /* XXX: use that in all the code */
ByteIOContext *pb;
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;
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;
int readonly; /* True if writing is prohibited to the file */
int conns_served;
int64_t bytes_served;
- int64_t feed_max_size; /* maximum storage size */
+ int64_t feed_max_size; /* maximum storage size, zero means unlimited */
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;
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 */
+static char logfilename[1024];
+static HTTPContext *first_http_ctx;
+static FFStream *first_feed; /* contains only feeds */
+static FFStream *first_stream; /* contains all streams, including feeds */
static void new_connection(int server_fd, int is_rtsp);
static void close_connection(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);
-static int compute_send_delay(HTTPContext *c);
/* RTSP handling */
static int rtsp_parse_request(HTTPContext *c);
static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h);
/* SDP handling */
-static int prepare_sdp_description(FFStream *stream, uint8_t **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);
-static int rtp_new_av_stream(HTTPContext *c,
- int stream_index, struct sockaddr_in *dest_addr);
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ 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,
+ HTTPContext *rtsp_c);
static const char *my_program_name;
static const char *my_program_dir;
static int no_launch;
static int need_to_start_children;
-int nb_max_connections;
-int nb_connections;
+static int nb_max_connections;
+static int nb_connections;
-int max_bandwidth;
-int current_bandwidth;
+static int max_bandwidth;
+static int current_bandwidth;
static long cur_time; // Making this global saves on passing it around everywhere
static FILE *logfile = NULL;
-static void http_log(const char *fmt, ...)
+static void __attribute__ ((format (printf, 1, 2))) http_log(const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
-
+
if (logfile) {
vfprintf(logfile, fmt, ap);
fflush(logfile);
{
char buf2[32];
- if (c->suppress_log)
+ 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,
+ http_log("%s - - [%s] \"%s %s %s\" %d %"PRId64"\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);
}
{
if (cur_time == drd->time1)
return 0;
-
- return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
-}
-static int get_longterm_datarate(DataRateData *drd, int64_t count)
-{
- /* You get the first 3 seconds flat out */
- if (cur_time - drd->time1 < 3000)
- return 0;
- return compute_datarate(drd, count);
+ return ((count - drd->count1) * 1000) / (cur_time - drd->time1);
}
perror ("socket");
return -1;
}
-
+
tmp = 1;
setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
close(server_fd);
return -1;
}
-
+
if (listen (server_fd, 5) < 0) {
perror ("listen");
close(server_fd);
for(stream = first_stream; stream != NULL; stream = stream->next) {
if (stream->is_multicast) {
/* open the RTP connection */
- snprintf(session_id, sizeof(session_id),
+ snprintf(session_id, sizeof(session_id),
"%08x%08x", (int)random(), (int)random());
/* choose a port if none given */
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;
}
if (open_input_stream(rtp_c, "") < 0) {
- fprintf(stderr, "Could not open input stream for stream '%s'\n",
+ 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;
+ for(stream_index = 0; stream_index < stream->nb_streams;
stream_index++) {
- dest_addr.sin_port = htons(stream->multicast_port +
+ 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",
+ 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);
}
rtsp_server_fd = socket_open_listen(&my_rtsp_addr);
if (rtsp_server_fd < 0)
return -1;
-
+
http_log("ffserver started.\n");
start_children(first_feed);
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 (need_to_start_children) {
HTTPContext *c = NULL;
len = sizeof(from_addr);
- fd = accept(server_fd, (struct sockaddr *)&from_addr,
+ fd = accept(server_fd, (struct sockaddr *)&from_addr,
&len);
if (fd < 0)
return;
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 = 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);
return;
}
}
+ /* 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);
/* 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);
+ 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)
+ if (c->stream)
nb_streams = c->stream->nb_streams;
-
+
for(i=0;i<nb_streams;i++) {
ctx = c->rtp_ctx[i];
if (ctx) {
/* prepare header */
if (url_open_dyn_buf(&ctx->pb) >= 0) {
av_write_trailer(ctx);
- (void) url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
}
}
}
- for(i=0; i<ctx->nb_streams; i++)
- av_free(ctx->streams[i]) ;
+ for(i=0; i<ctx->nb_streams; i++)
+ av_free(ctx->streams[i]) ;
- if (c->stream)
+ if (c->stream && !c->post && c->stream->stream_type == STREAM_TYPE_LIVE)
current_bandwidth -= c->stream->bandwidth;
av_freep(&c->pb_buffer);
+ av_freep(&c->packet_buffer);
av_free(c->buffer);
av_free(c);
nb_connections--;
static int handle_connection(HTTPContext *c)
{
int len, ret;
-
+
switch(c->state) {
case HTTPSTATE_WAIT_REQUEST:
case RTSPSTATE_WAIT_REQUEST:
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 */
+ /* search for end of request. */
uint8_t *ptr;
c->buffer_ptr += len;
ptr = c->buffer_ptr;
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
return -1;
- }
+ } else goto read_loop;
}
break;
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;
/* 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;
int best = -1;
for (i = 0; i < feed->nb_streams; i++) {
- AVCodecContext *feed_codec = &feed->streams[i]->codec;
+ AVCodecContext *feed_codec = feed->streams[i]->codec;
if (feed_codec->codec_id != codec->codec_id ||
feed_codec->sample_rate != codec->sample_rate ||
/* Potential stream */
- /* We want the fastest stream less than bit_rate, or the slowest
+ /* We want the fastest stream less than bit_rate, or the slowest
* faster than bit_rate
*/
return 0;
for (i = 0; i < req->nb_streams; i++) {
- AVCodecContext *codec = &req->streams[i]->codec;
+ AVCodecContext *codec = req->streams[i]->codec;
switch(rates[i]) {
case 0:
static void do_switch_stream(HTTPContext *c, int i)
{
if (c->switch_feed_streams[i] >= 0) {
-#ifdef PHILIP
+#ifdef PHILIP
c->feed_streams[i] = c->switch_feed_streams[i];
#endif
static int http_parse_request(HTTPContext *c)
{
char *p;
- int post;
enum RedirType redir_type;
char cmd[32];
char info[1024], *filename;
pstrcpy(c->method, sizeof(c->method), cmd);
if (!strcmp(cmd, "GET"))
- post = 0;
+ c->post = 0;
else if (!strcmp(cmd, "POST"))
- post = 1;
+ c->post = 1;
else
return -1;
return -1;
pstrcpy(c->protocol, sizeof(c->protocol), protocol);
-
+
+ if (ffserver_debug)
+ http_log("New connection: %s %s\n", cmd, url);
+
/* find the filename and the optional info string in the request */
p = url;
if (*p == '/')
redir_type = REDIR_SDP;
compute_real_filename(filename, sizeof(url) - 1);
}
-
+
stream = first_stream;
while (stream != NULL) {
if (!strcmp(stream->filename, filename) && validate_acl(stream, c))
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 += sprintf(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 += snprintf(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 && stream->stream_type == STREAM_TYPE_LIVE) {
+ if (c->post == 0 && stream->stream_type == STREAM_TYPE_LIVE) {
current_bandwidth += stream->bandwidth;
}
-
- if (post == 0 && max_bandwidth < current_bandwidth) {
+
+ if (c->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",
+ 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 += snprintf(q, q - (char *) c->buffer + c->buffer_size, "<p>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, "<p>The bandwidth being served (including your stream) is %dkbit/sec, and this exceeds the limit of %dkbit/sec.</p>\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;
c->state = HTTPSTATE_SEND_HEADER;
return 0;
}
-
+
if (redir_type != REDIR_NONE) {
char *hostinfo = 0;
-
+
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
if (strncasecmp(p, "Host:", 5) == 0) {
hostinfo = p + 5;
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 += 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",
+ 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 += snprintf(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",
- hostname, ntohs(my_rtsp_addr.sin_port),
+ 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;
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);
-
+
/* XXX: should use a dynamic buffer */
- sdp_data_size = prepare_sdp_description(stream,
- &sdp_data,
+ 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);
}
}
- sprintf(msg, "ASX/RAM file not handled");
+ snprintf(msg, sizeof(msg), "ASX/RAM file not handled");
goto send_error;
}
/* XXX: add there authenticate and IP match */
- if (post) {
+ if (c->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
*/
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;
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);
+ 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;
c->buffer_ptr = c->buffer;
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;
for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++) {
}
- url_fprintf(pb, "%lld%c", count, *s);
+ url_fprintf(pb, "%"PRId64"%c", count, *s);
}
static void compute_stats(HTTPContext *c)
url_fprintf(pb, "Content-type: %s\r\n", "text/html");
url_fprintf(pb, "Pragma: no-cache\r\n");
url_fprintf(pb, "\r\n");
-
+
url_fprintf(pb, "<HEAD><TITLE>FFServer Status</TITLE>\n");
if (c->stream->feed_filename) {
url_fprintf(pb, "<link rel=\"shortcut icon\" href=\"%s\">\n", c->stream->feed_filename);
strcpy(eosf - 4, ".asx");
} else if (strcmp(eosf - 3, ".rm") == 0) {
strcpy(eosf - 3, ".ram");
- } else if (stream->fmt == &rtp_mux) {
+ } else if (stream->fmt == &rtp_muxer) {
/* generate a sample RTSP director if
unicast. Generate an SDP redirector if
multicast */
strcpy(eosf, ".rtsp");
}
}
-
- url_fprintf(pb, "<TR><TD><A HREF=\"/%s\">%s</A> ",
+
+ url_fprintf(pb, "<TR><TD><A HREF=\"/%s\">%s</A> ",
sfilename, stream->filename);
url_fprintf(pb, "<td align=right> %d <td align=right> ",
stream->conns_served);
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) {
+ 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;
+ audio_bit_rate += st->codec->bit_rate;
if (codec) {
if (*audio_codec_name)
audio_codec_name_extra = "...";
}
break;
case CODEC_TYPE_VIDEO:
- video_bit_rate += st->codec.bit_rate;
+ video_bit_rate += st->codec->bit_rate;
if (codec) {
if (*video_codec_name)
video_codec_name_extra = "...";
video_codec_name = codec->name;
}
break;
+ case CODEC_TYPE_DATA:
+ video_bit_rate += st->codec->bit_rate;
+ break;
default:
av_abort();
}
}
- 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",
+ 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,
char ps_cmd[64];
/* This is somewhat linux specific I guess */
- snprintf(ps_cmd, sizeof(ps_cmd),
- "ps -o \"%%cpu,cputime\" --no-headers %d",
+ 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,
+
+ 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);
for (i = 0; i < stream->nb_streams; i++) {
AVStream *st = stream->streams[i];
- AVCodec *codec = avcodec_find_encoder(st->codec.codec_id);
+ AVCodec *codec = avcodec_find_encoder(st->codec->codec_id);
const char *type = "unknown";
char parameters[64];
parameters[0] = 0;
- switch(st->codec.codec_type) {
+ 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 / st->codec.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();
}
url_fprintf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d<td>%s<td>%s\n",
- i, type, st->codec.bit_rate/1000, codec ? codec->name : "", parameters);
+ i, type, st->codec->bit_rate/1000, codec ? codec->name : "", parameters);
}
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) {
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
FeedData *fdata = st->priv_data;
- enc = &st->codec;
-
+ 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;
- url_fprintf(pb, "<TR><TD>%s <TD> %d <TD> %Ld <TD> %0.1f\n",
+ url_fprintf(pb, "<TR><TD>%s <TD> %d <TD> %"PRId64" <TD> %0.1f\n",
buf, enc->frame_number, fdata->data_count, avg / 1000.0);
}
url_fprintf(pb, "</TABLE>\n");
if (c1->stream) {
for (j = 0; j < c1->stream->nb_streams; j++) {
if (!c1->stream->feed) {
- bitrate += c1->stream->streams[j]->codec.bit_rate;
+ 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;
+ bitrate += c1->stream->feed->streams[c1->feed_streams[j]]->codec->bit_rate;
}
}
}
i++;
p = inet_ntoa(c1->from_addr.sin_addr);
- 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 : "",
+ 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,
+ p,
c1->protocol,
http_state[c1->state]);
fmt_bytecount(pb, bitrate);
c1 = c1->next;
}
url_fprintf(pb, "</TABLE>\n");
-
+
/* date */
ti = time(NULL);
p = ctime(&ti);
AVStream *st = s->streams[i];
AVCodec *codec;
- if (!st->codec.codec) {
- codec = avcodec_find_decoder(st->codec.codec_id);
+ 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;
+ st->codec->parse_only = 1;
+ if (avcodec_open(st->codec, codec) < 0) {
+ st->codec->parse_only = 0;
}
}
}
#if 0
{ time_t when = stream_pos / 1000000;
- http_log("Stream pos = %lld, time=%s", stream_pos, ctime(&when));
+ http_log("Stream pos = %"PRId64", time=%s", stream_pos, ctime(&when));
}
#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;
}
c->fmt_in = s;
-
+
/* open each parser */
for(i=0;i<s->nb_streams;i++)
open_parser(s, i);
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) {
+ if (c->pts_stream_index == 0 &&
+ c->stream->streams[i]->codec->codec_type == CODEC_TYPE_VIDEO) {
c->pts_stream_index = i;
}
}
+#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)
-{
- 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_t)s->pts_den;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.sample_rate);
- break;
- case CODEC_TYPE_VIDEO:
- st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)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_t)s->pts_den * st->codec.frame_size;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)s->pts_num * st->codec.sample_rate);
- break;
- case CODEC_TYPE_VIDEO:
- st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
- av_frac_init(&st->pts, st->pts.val, 0,
- (int64_t)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;
- }
- }
- }
-}
-
-static int compute_send_delay(HTTPContext *c)
+/* return the server clock (in us) */
+static int64_t get_server_clock(HTTPContext *c)
{
- int64_t cur_pts, delta_pts, next_pts;
- int delay1;
-
/* compute current pts value from system time */
- cur_pts = ((int64_t)(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;
+ return (int64_t)(cur_time - c->start_time) * 1000LL;
}
-#else
-/* just fall backs */
-static int av_read_frame(AVFormatContext *s, AVPacket *pkt)
+/* return the estimated time at which the current packet must be sent
+ (in us) */
+static int64_t get_packet_send_clock(HTTPContext *c)
{
- return av_read_packet(s, pkt);
-}
+ int bytes_left, bytes_sent, frame_bytes;
-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;
+ frame_bytes = c->cur_frame_bytes;
+ if (frame_bytes <= 0) {
+ return c->cur_pts;
+ } else {
+ 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 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));
- pstrcpy(c->fmt_ctx.author, sizeof(c->fmt_ctx.author),
+ pstrcpy(c->fmt_ctx.author, sizeof(c->fmt_ctx.author),
c->stream->author);
- pstrcpy(c->fmt_ctx.comment, sizeof(c->fmt_ctx.comment),
+ pstrcpy(c->fmt_ctx.comment, sizeof(c->fmt_ctx.comment),
c->stream->comment);
- pstrcpy(c->fmt_ctx.copyright, sizeof(c->fmt_ctx.copyright),
+ pstrcpy(c->fmt_ctx.copyright, sizeof(c->fmt_ctx.copyright),
c->stream->copyright);
- pstrcpy(c->fmt_ctx.title, sizeof(c->fmt_ctx.title),
+ pstrcpy(c->fmt_ctx.title, sizeof(c->fmt_ctx.title),
c->stream->title);
/* open output stream by using specified codecs */
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));
+ st->codec= avcodec_alloc_context();
c->fmt_ctx.streams[i] = st;
/* if file or feed, then just take streams from FFStream struct */
- if (!c->stream->feed ||
+ 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));
- st->codec.frame_number = 0; /* XXX: should be done in
+ 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;
+ st->codec->coded_frame = &dummy_frame;
}
c->got_key_frame = 0;
/* find a new packet */
{
AVPacket pkt;
-
+
/* read a packet from the input stream */
if (c->stream->feed) {
- ffm_set_write_index(c->fmt_in,
+ ffm_set_write_index(c->fmt_in,
c->stream->feed->feed_write_index,
c->stream->feed->feed_size);
}
- if (c->stream->max_time &&
+ 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;
} 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 */
if (pkt.flags & PKT_FLAG_KEY) {
c->got_key_frame |= 1 << i;
}
- /* See if we have all the key frames, then
+ /* 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
+ * 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).
+ * audio streams (for which every frame is
+ * typically a key frame).
*/
- if (!c->stream->send_on_key ||
+ if (!c->stream->send_on_key ||
((c->got_key_frame + 1) >> c->stream->nb_streams)) {
goto send_it;
}
}
} else {
AVCodecContext *codec;
-
+
send_it:
/* specific handling for RTP: we use several
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];
- codec = &ctx->streams[0]->codec;
+ if(!ctx) {
+ av_free_packet(&pkt);
+ break;
+ }
+ codec = ctx->streams[0]->codec;
/* only one stream per RTP connection */
pkt.stream_index = 0;
} else {
ctx = &c->fmt_ctx;
/* Fudge here */
- codec = &ctx->streams[pkt.stream_index]->codec;
+ codec = ctx->streams[pkt.stream_index]->codec;
}
-
+
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_t)1000000 >=
- (SHORT_TERM_BANDWIDTH / 8) * (int64_t)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 (fd < 0)
return -1;
c->feed_fd = fd;
-
+
c->stream->feed_write_index = ffm_read_write_index(fd);
c->stream->feed_size = lseek(fd, 0, SEEK_END);
lseek(fd, 0, SEEK_SET);
c->stream->feed_opened = 1;
return 0;
}
-
+
static int http_receive_data(HTTPContext *c)
{
HTTPContext *c1;
}
}
+ 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 header */
if (c->data_count > FFM_PACKET_SIZE) {
-
- // printf("writing pos=0x%Lx size=0x%Lx\n", feed->feed_write_index, feed->feed_size);
+
+ // printf("writing pos=0x%"PRIx64" size=0x%"PRIx64"\n", feed->feed_write_index, feed->feed_size);
/* XXX: use llseek or url_seek */
lseek(c->feed_fd, feed->feed_write_index, SEEK_SET);
write(c->feed_fd, c->buffer, FFM_PACKET_SIZE);
-
+
feed->feed_write_index += FFM_PACKET_SIZE;
/* update file size */
if (feed->feed_write_index > c->stream->feed_size)
feed->feed_size = feed->feed_write_index;
/* handle wrap around if max file size reached */
- if (feed->feed_write_index >= c->stream->feed_max_size)
+ if (c->stream->feed_max_size && feed->feed_write_index >= c->stream->feed_max_size)
feed->feed_write_index = FFM_PACKET_SIZE;
/* write index */
/* wake up any waiting connections */
for(c1 = first_http_ctx; c1 != NULL; c1 = c1->next) {
- if (c1->state == HTTPSTATE_WAIT_FEED &&
+ if (c1->state == HTTPSTATE_WAIT_FEED &&
c1->stream->feed == c->stream->feed) {
c1->state = HTTPSTATE_SEND_DATA;
}
s.priv_data = av_mallocz(fmt_in->priv_data_size);
if (!s.priv_data)
goto fail;
- } else
- s.priv_data = NULL;
+ } else
+ s.priv_data = NULL;
if (fmt_in->read_header(&s, 0) < 0) {
av_freep(&s.priv_data);
goto fail;
}
for (i = 0; i < s.nb_streams; i++) {
- memcpy(&feed->streams[i]->codec,
- &s.streams[i]->codec, sizeof(AVCodecContext));
- }
+ memcpy(feed->streams[i]->codec,
+ s.streams[i]->codec, sizeof(AVCodecContext));
+ }
av_freep(&s.priv_data);
}
c->buffer_ptr = c->buffer;
char buf2[32];
switch(error_number) {
-#define DEF(n, c, s) case c: str = s; break;
+#define DEF(n, c, s) case c: str = s; break;
#include "rtspcodes.h"
#undef DEF
default:
str = "Unknown Error";
break;
}
-
+
url_fprintf(c->pb, "RTSP/1.0 %d %s\r\n", error_number, str);
url_fprintf(c->pb, "CSeq: %d\r\n", c->seq);
ByteIOContext pb1;
int len;
RTSPHeader header1, *header = &header1;
-
+
c->buffer_ptr[0] = '\0';
p = c->buffer;
-
+
get_word(cmd, sizeof(cmd), &p);
get_word(url, sizeof(url), &p);
get_word(protocol, sizeof(protocol), &p);
/* XXX: move that to rtsp.c, but would need to replace FFStream by
AVFormatContext */
-static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
+static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
struct in_addr my_ip)
{
ByteIOContext pb1, *pb = &pb1;
int i, payload_type, port, private_payload_type, j;
const char *ipstr, *title, *mediatype;
AVStream *st;
-
+
if (url_open_dyn_buf(pb) < 0)
return -1;
-
+
/* general media info */
url_fprintf(pb, "v=0\n");
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 */
- payload_type = rtp_get_payload_type(&st->codec);
+ payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = private_payload_type++;
if (stream->is_multicast) {
} else {
port = 0;
}
- url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
+ 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) {
+ switch(st->codec->codec_id) {
case CODEC_ID_MPEG4:
{
uint8_t *data;
- url_fprintf(pb, "a=rtpmap:%d MP4V-ES/%d\n",
+ url_fprintf(pb, "a=rtpmap:%d MP4V-ES/%d\n",
payload_type, 90000);
/* we must also add the mpeg4 header */
- data = st->codec.extradata;
+ data = st->codec->extradata;
if (data) {
- url_fprintf(pb, "a=fmtp:%d config=");
- for(j=0;j<st->codec.extradata_size;j++) {
+ url_fprintf(pb, "a=fmtp:%d config=", payload_type);
+ for(j=0;j<st->codec->extradata_size;j++) {
url_fprintf(pb, "%02x", data[j]);
}
url_fprintf(pb, "\n");
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++;
for(stream = first_stream; stream != NULL; stream = stream->next) {
- if (!stream->is_feed && stream->fmt == &rtp_mux &&
+ if (!stream->is_feed && stream->fmt == &rtp_muxer &&
!strcmp(path, stream->filename)) {
goto found;
}
/* 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;
}
-static void rtsp_cmd_setup(HTTPContext *c, const char *url,
+static void rtsp_cmd_setup(HTTPContext *c, const char *url,
RTSPHeader *h)
{
FFStream *stream;
RTSPTransportField *th;
struct sockaddr_in dest_addr;
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++;
/* now check each stream */
for(stream = first_stream; stream != NULL; stream = stream->next) {
- if (!stream->is_feed && stream->fmt == &rtp_mux) {
+ if (!stream->is_feed && stream->fmt == &rtp_muxer) {
/* accept aggregate filenames only if single stream */
if (!strcmp(path, stream->filename)) {
if (stream->nb_streams != 1) {
stream_index = 0;
goto found;
}
-
+
for(stream_index = 0; stream_index < stream->nb_streams;
stream_index++) {
- snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
stream->filename, stream_index);
if (!strcmp(path, buf))
goto found;
/* generate session id if needed */
if (h->session_id[0] == '\0') {
- snprintf(h->session_id, sizeof(h->session_id),
+ snprintf(h->session_id, sizeof(h->session_id),
"%08x%08x", (int)random(), (int)random());
}
/* 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
to be done for a given file) */
if (rtp_c->stream != stream) {
rtsp_reply_error(c, RTSP_STATUS_SERVICE);
return;
}
-
+
/* test if stream is already set up */
if (rtp_c->rtp_ctx[stream_index]) {
rtsp_reply_error(c, RTSP_STATUS_STATE);
/* check transport */
th = find_transport(h, rtp_c->rtp_protocol);
- if (!th || (th->protocol == RTSP_PROTOCOL_RTP_UDP &&
+ if (!th || (th->protocol == RTSP_PROTOCOL_RTP_UDP &&
th->client_port_min <= 0)) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
setup.transport_option[0] = '\0';
dest_addr = rtp_c->from_addr;
dest_addr.sin_port = htons(th->client_port_min);
-
+
/* add transport option if needed */
if (ff_rtsp_callback) {
setup.ipaddr = ntohl(dest_addr.sin_addr.s_addr);
- if (ff_rtsp_callback(RTSP_ACTION_SERVER_SETUP, rtp_c->session_id,
+ if (ff_rtsp_callback(RTSP_ACTION_SERVER_SETUP, rtp_c->session_id,
(char *)&setup, sizeof(setup),
stream->rtsp_option) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
}
dest_addr.sin_addr.s_addr = htonl(setup.ipaddr);
}
-
+
/* 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;
}
url_fprintf(c->pb, ";%s", setup.transport_option);
}
url_fprintf(c->pb, "\r\n");
-
+
url_fprintf(c->pb, "\r\n");
}
/* find an rtp connection by using the session ID. Check consistency
with filename */
-static HTTPContext *find_rtp_session_with_url(const char *url,
+static HTTPContext *find_rtp_session_with_url(const char *url,
const char *session_id)
{
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)
rtsp_reply_error(c, RTSP_STATUS_SESSION);
return;
}
-
+
if (rtp_c->state != HTTPSTATE_SEND_DATA &&
rtp_c->state != HTTPSTATE_WAIT_FEED &&
rtp_c->state != HTTPSTATE_READY) {
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 */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
rtsp_reply_error(c, RTSP_STATUS_SESSION);
return;
}
-
+
if (rtp_c->state != HTTPSTATE_SEND_DATA &&
rtp_c->state != HTTPSTATE_WAIT_FEED) {
rtsp_reply_error(c, RTSP_STATUS_STATE);
return;
}
-
+
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 */
rtsp_reply_error(c, RTSP_STATUS_SESSION);
return;
}
-
+
/* abort the session */
close_connection(rtp_c);
if (ff_rtsp_callback) {
- ff_rtsp_callback(RTSP_ACTION_SERVER_TEARDOWN, rtp_c->session_id,
+ ff_rtsp_callback(RTSP_ACTION_SERVER_TEARDOWN, rtp_c->session_id,
NULL, 0,
rtp_c->stream->rtsp_option);
}
/********************************************************************/
/* RTP handling */
-static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
- FFStream *stream, const char *session_id)
+static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
+ 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)
goto fail;
-
+
/* add a new connection */
c = av_mallocz(sizeof(HTTPContext));
if (!c)
goto fail;
-
+
c->fd = -1;
c->poll_entry = NULL;
c->from_addr = *from_addr;
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;
c->next = first_http_ctx;
first_http_ctx = c;
return c;
-
+
fail:
if (c) {
av_free(c->buffer);
}
/* 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)
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr,
+ HTTPContext *rtsp_c)
{
AVFormatContext *ctx;
AVStream *st;
URLContext *h;
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;
+ ctx->oformat = &rtp_muxer;
st = av_mallocz(sizeof(AVStream));
if (!st)
goto fail;
+ st->codec= avcodec_alloc_context();
ctx->nb_streams = 1;
ctx->streams[0] = st;
- if (!c->stream->feed ||
+ if (!c->stream->feed ||
c->stream->feed == c->stream) {
memcpy(st, c->stream->streams[stream_index], sizeof(AVStream));
} else {
- memcpy(st,
+ memcpy(st,
c->stream->feed->streams[c->stream->feed_streams[stream_index]],
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) {
int ttl;
if (!ttl)
ttl = 16;
snprintf(ctx->filename, sizeof(ctx->filename),
- "rtp://%s:%d?multicast=1&ttl=%d",
+ "rtp://%s:%d?multicast=1&ttl=%d",
ipaddr, ntohs(dest_addr->sin_port), ttl);
} else {
snprintf(ctx->filename, sizeof(ctx->filename),
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",
- ipaddr, ntohs(dest_addr->sin_port),
- ctime1(buf2),
- c->stream->filename, stream_index);
+ 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->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;
}
}
url_close_dyn_buf(&ctx->pb, &dummy_buf);
av_free(dummy_buf);
-
+
c->rtp_ctx[stream_index] = ctx;
return 0;
}
fst = av_mallocz(sizeof(AVStream));
if (!fst)
return NULL;
+ fst->codec= avcodec_alloc_context();
fst->priv_data = av_mallocz(sizeof(FeedData));
- memcpy(&fst->codec, codec, sizeof(AVCodecContext));
- fst->codec.coded_frame = &dummy_frame;
+ 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;
}
AVCodecContext *av, *av1;
int i;
- av = &st->codec;
+ av = st->codec;
for(i=0;i<feed->nb_streams;i++) {
st = feed->streams[i];
- av1 = &st->codec;
+ av1 = st->codec;
if (av1->codec_id == av->codec_id &&
av1->codec_type == av->codec_type &&
av1->bit_rate == av->bit_rate) {
case CODEC_TYPE_VIDEO:
if (av1->width == av->width &&
av1->height == av->height &&
- av1->frame_rate == av->frame_rate &&
- av1->frame_rate_base == av->frame_rate_base &&
+ 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;
}
}
}
-
+
fst = add_av_stream1(feed, av);
if (!fst)
return -1;
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) {
+ if (st->codec->codec_id == CODEC_ID_MPEG4 &&
+ 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) {
+ if (st->codec->codec_id == CODEC_ID_MPEG4 &&
+ st->codec->extradata_size == 0) {
+ av_freep(&st->codec->extradata);
/* fill extradata with the header */
/* XXX: we make hard suppositions here ! */
p = pkt.data;
while (p < pkt.data + pkt.size - 4) {
/* stop when vop header is found */
- if (p[0] == 0x00 && p[1] == 0x00 &&
+ if (p[0] == 0x00 && p[1] == 0x00 &&
p[2] == 0x01 && p[3] == 0xb6) {
size = p - pkt.data;
// av_hex_dump(pkt.data, size);
- st->codec.extradata = av_malloc(size);
- st->codec.extradata_size = size;
- memcpy(st->codec.extradata, pkt.data, size);
+ st->codec->extradata = av_malloc(size);
+ st->codec->extradata_size = size;
+ memcpy(st->codec->extradata, pkt.data, size);
break;
}
p++;
/* the stream comes from a file */
/* try to open the file */
/* open stream */
- if (av_open_input_file(&infile, stream->feed_filename,
- NULL, 0, NULL) < 0) {
+ stream->ap_in = av_mallocz(sizeof(AVFormatParameters));
+ if (stream->fmt == &rtp_muxer) {
+ /* 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,
+ 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:
/* find all the AVStreams inside and reference them in
'stream' */
if (av_find_stream_info(infile) < 0) {
- http_log("Could not find codec parameters from '%s'",
+ http_log("Could not find codec parameters from '%s'",
stream->feed_filename);
av_close_input_file(infile);
goto fail;
extract_mpeg4_header(infile);
for(i=0;i<infile->nb_streams;i++) {
- add_av_stream1(stream, &infile->streams[i]->codec);
+ add_av_stream1(stream, infile->streams[i]->codec);
}
av_close_input_file(infile);
}
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;
- ccf = &sf->codec;
- ccs = &ss->codec;
+ ccf = sf->codec;
+ ccs = ss->codec;
#define CHECK_CODEC(x) (ccf->x != ccs->x)
if (CHECK_CODEC(codec) || CHECK_CODEC(codec_type)) {
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) ||
- CHECK_CODEC(frame_rate_base) ||
+ 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);
feed->feed_write_index = ffm_read_write_index(fd);
feed->feed_size = lseek(fd, 0, SEEK_END);
/* ensure that we do not wrap before the end of file */
- if (feed->feed_max_size < feed->feed_size)
+ if (feed->feed_max_size && feed->feed_max_size < feed->feed_size)
feed->feed_max_size = feed->feed_size;
close(fd);
{
int bandwidth, i;
FFStream *stream;
-
+
for(stream = first_stream; stream != NULL; stream = stream->next) {
bandwidth = 0;
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
- switch(st->codec.codec_type) {
+ switch(st->codec->codec_type) {
case CODEC_TYPE_AUDIO:
case CODEC_TYPE_VIDEO:
- bandwidth += st->codec.bit_rate;
+ bandwidth += st->codec->bit_rate;
break;
default:
break;
case CODEC_TYPE_VIDEO:
if (av->bit_rate == 0)
av->bit_rate = 64000;
- if (av->frame_rate == 0){
- av->frame_rate = 5;
- av->frame_rate_base = 1;
+ 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->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();
st = av_mallocz(sizeof(AVStream));
if (!st)
return;
+ st->codec = avcodec_alloc_context();
stream->streams[stream->nb_streams++] = st;
- memcpy(&st->codec, av, sizeof(AVCodecContext));
+ memcpy(st->codec, av, sizeof(AVCodecContext));
}
static int opt_audio_codec(const char *arg)
/* simplistic plugin support */
-#ifdef CONFIG_HAVE_DLOPEN
-void load_module(const char *filename)
+#ifdef HAVE_DLOPEN
+static void load_module(const char *filename)
{
void *dll;
void (*init_func)(void);
filename, dlerror());
return;
}
-
+
init_func = dlsym(dll, "ffserver_module_init");
if (!init_func) {
- fprintf(stderr,
+ fprintf(stderr,
"%s: init function 'ffserver_module_init()' not found\n",
filename);
dlclose(dll);
perror(filename);
return -1;
}
-
+
errors = 0;
line_num = 0;
first_stream = NULL;
break;
line_num++;
p = line;
- while (isspace(*p))
+ while (isspace(*p))
p++;
if (*p == '\0' || *p == '#')
continue;
get_arg(cmd, sizeof(cmd), &p);
-
+
if (!strcasecmp(cmd, "Port")) {
get_arg(arg, sizeof(arg), &p);
my_http_addr.sin_port = htons (atoi(arg));
} else if (!strcasecmp(cmd, "BindAddress")) {
get_arg(arg, sizeof(arg), &p);
if (!inet_aton(arg, &my_http_addr.sin_addr)) {
- fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
+ fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
filename, line_num, arg);
errors++;
}
} else if (!strcasecmp(cmd, "RTSPBindAddress")) {
get_arg(arg, sizeof(arg), &p);
if (!inet_aton(arg, &my_rtsp_addr.sin_addr)) {
- fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
+ fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
filename, line_num, arg);
errors++;
}
get_arg(arg, sizeof(arg), &p);
val = atoi(arg);
if (val < 1 || val > HTTP_MAX_CONNECTIONS) {
- fprintf(stderr, "%s:%d: Invalid MaxClients: %s\n",
+ fprintf(stderr, "%s:%d: Invalid MaxClients: %s\n",
filename, line_num, arg);
errors++;
} else {
get_arg(arg, sizeof(arg), &p);
val = atoi(arg);
if (val < 10 || val > 100000) {
- fprintf(stderr, "%s:%d: Invalid MaxBandwidth: %s\n",
+ fprintf(stderr, "%s:%d: Invalid MaxBandwidth: %s\n",
filename, line_num, arg);
errors++;
} else {
/* add in feed list */
*last_feed = feed;
last_feed = &feed->next_feed;
-
+
get_arg(feed->filename, sizeof(feed->filename), &p);
q = strrchr(feed->filename, '>');
if (*q)
feed->child_argv = (char **) av_mallocz(64 * sizeof(char *));
- feed->child_argv[0] = av_malloc(7);
- strcpy(feed->child_argv[0], "ffmpeg");
-
- for (i = 1; i < 62; i++) {
+ for (i = 0; i < 62; i++) {
char argbuf[256];
get_arg(argbuf, sizeof(argbuf), &p);
if (!argbuf[0])
break;
- if (strlen(argbuf + 1)) {
- feed->child_argv[i] = av_malloc(strlen(argbuf + 1));
- strcpy(feed->child_argv[i], argbuf);
- } else
- feed->child_argv[i] = NULL;
+ feed->child_argv[i] = av_malloc(strlen(argbuf) + 1);
+ strcpy(feed->child_argv[i], argbuf);
}
feed->child_argv[i] = av_malloc(30 + strlen(feed->filename));
- snprintf(feed->child_argv[i], 256, "http://127.0.0.1:%d/%s",
+ snprintf(feed->child_argv[i], 30+strlen(feed->filename),
+ "http://%s:%d/%s",
+ (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" :
+ inet_ntoa(my_http_addr.sin_addr),
ntohs(my_http_addr.sin_port), feed->filename);
+
+ if (ffserver_debug)
+ {
+ int j;
+ fprintf(stdout, "Launch commandline: ");
+ for (j = 0; j <= i; j++)
+ fprintf(stdout, "%s ", feed->child_argv[j]);
+ fprintf(stdout, "\n");
+ }
}
} else if (!strcasecmp(cmd, "ReadOnlyFile")) {
if (feed) {
#if 0
} else {
/* Make sure that we start out clean */
- if (unlink(feed->feed_filename) < 0
+ if (unlink(feed->feed_filename) < 0
&& errno != ENOENT) {
fprintf(stderr, "%s:%d: Unable to clean old feed file '%s': %s\n",
filename, line_num, feed->feed_filename, strerror(errno));
get_arg(arg, sizeof(arg), &p);
if (stream) {
FFStream *sfeed;
-
+
sfeed = first_feed;
while (sfeed != NULL) {
if (!strcmp(sfeed->filename, arg))
stream->stream_type = STREAM_TYPE_LIVE;
/* jpeg cannot be used here, so use single frame jpeg */
if (!strcmp(arg, "jpeg"))
- strcpy(arg, "singlejpeg");
+ strcpy(arg, "mjpeg");
stream->fmt = guess_stream_format(arg, NULL, NULL);
if (!stream->fmt) {
- fprintf(stderr, "%s:%d: Unknown Format: %s\n",
+ fprintf(stderr, "%s:%d: Unknown Format: %s\n",
filename, line_num, arg);
errors++;
}
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);
} else {
- fprintf(stderr, "%s:%d: FaviconURL only permitted for status streams\n",
+ fprintf(stderr, "%s:%d: FaviconURL only permitted for status streams\n",
filename, line_num);
errors++;
}
get_arg(arg, sizeof(arg), &p);
audio_id = opt_audio_codec(arg);
if (audio_id == CODEC_ID_NONE) {
- fprintf(stderr, "%s:%d: Unknown AudioCodec: %s\n",
+ fprintf(stderr, "%s:%d: Unknown AudioCodec: %s\n",
filename, line_num, arg);
errors++;
}
get_arg(arg, sizeof(arg), &p);
video_id = opt_video_codec(arg);
if (video_id == CODEC_ID_NONE) {
- fprintf(stderr, "%s:%d: Unknown VideoCodec: %s\n",
+ fprintf(stderr, "%s:%d: Unknown VideoCodec: %s\n",
filename, line_num, arg);
errors++;
}
if (stream) {
audio_enc.sample_rate = atoi(arg);
}
- } else if (!strcasecmp(cmd, "AudioQuality")) {
- get_arg(arg, sizeof(arg), &p);
+ } else if (!strcasecmp(cmd, "AudioQuality")) {
+ get_arg(arg, sizeof(arg), &p);
if (stream) {
// audio_enc.quality = atof(arg) * 1000;
}
video_enc.rc_min_rate = minrate * 1000;
video_enc.rc_max_rate = maxrate * 1000;
} else {
- fprintf(stderr, "%s:%d: Incorrect format for VideoBitRateRange -- should be <min>-<max>: %s\n",
+ fprintf(stderr, "%s:%d: Incorrect format for VideoBitRateRange -- should be <min>-<max>: %s\n",
filename, line_num, arg);
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_base= DEFAULT_FRAME_RATE_BASE;
- video_enc.frame_rate = (int)(strtod(arg, NULL) * video_enc.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.flags |= CODEC_FLAG_HQ;
+ video_enc.mb_decision = FF_MB_DECISION_BITS; //FIXME remove
video_enc.flags |= CODEC_FLAG_4MV;
}
+ } else if (!strcasecmp(cmd, "BitExact")) {
+ if (stream) {
+ video_enc.flags |= CODEC_FLAG_BITEXACT;
+ }
+ } else if (!strcasecmp(cmd, "DctFastint")) {
+ if (stream) {
+ video_enc.dct_algo = FF_DCT_FASTINT;
+ }
+ } else if (!strcasecmp(cmd, "IdctSimple")) {
+ if (stream) {
+ video_enc.idct_algo = FF_IDCT_SIMPLE;
+ }
+ } else if (!strcasecmp(cmd, "Qscale")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ video_enc.flags |= CODEC_FLAG_QSCALE;
+ video_enc.global_quality = FF_QP2LAMBDA * atoi(arg);
+ }
} else if (!strcasecmp(cmd, "VideoQDiff")) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
get_arg(arg, sizeof(arg), &p);
if (stream) {
if (!inet_aton(arg, &stream->multicast_ip)) {
- fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
+ fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
filename, line_num, arg);
errors++;
}
redirect = NULL;
} else if (!strcasecmp(cmd, "LoadModule")) {
get_arg(arg, sizeof(arg), &p);
-#ifdef CONFIG_HAVE_DLOPEN
+#ifdef HAVE_DLOPEN
load_module(arg);
#else
- fprintf(stderr, "%s:%d: Module support not compiled into this version: '%s'\n",
+ fprintf(stderr, "%s:%d: Module support not compiled into this version: '%s'\n",
filename, line_num, arg);
errors++;
#endif
} else {
- fprintf(stderr, "%s:%d: Incorrect keyword: '%s'\n",
+ fprintf(stderr, "%s:%d: Incorrect keyword: '%s'\n",
filename, line_num, cmd);
errors++;
}
}
#endif
-static void help(void)
+static void show_banner(void)
+{
+ printf("ffserver version " FFMPEG_VERSION ", Copyright (c) 2000-2006 Fabrice Bellard, et al.\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"
);
}
-static 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"
+ "FFmpeg 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"
- "version 2 of the License, or (at your option) any later version.\n"
+ "version 2.1 of the License, or (at your option) any later version.\n"
"\n"
- "This library is distributed in the hope that it will be useful,\n"
+ "FFmpeg is distributed in the hope that it will be useful,\n"
"but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
"MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU\n"
"Lesser General Public License for more details.\n"
"\n"
"You should have received a copy of the GNU Lesser General Public\n"
- "License along with this library; if not, write to the Free Software\n"
- "Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA\n"
+ "License along with FFmpeg; if not, write to the Free Software\n"
+ "Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA\n"
);
}
if (uptime < 30) {
/* Turn off any more restarts */
feed->child_argv = 0;
- }
+ }
}
}
}
my_program_name = argv[0];
my_program_dir = getcwd(0, 0);
ffserver_daemon = 1;
-
+
for(;;) {
c = getopt(argc, argv, "ndLh?f:");
if (c == -1)
break;
switch(c) {
case 'L':
- licence();
+ show_license();
exit(1);
case '?':
case 'h':
- help();
+ show_help();
exit(1);
case 'n':
no_launch = 1;
my_rtsp_addr.sin_family = AF_INET;
my_rtsp_addr.sin_port = htons (5454);
my_rtsp_addr.sin_addr.s_addr = htonl (INADDR_ANY);
-
+
nb_max_connections = 5;
max_bandwidth = 1000;
first_stream = NULL;