[SCM] libav/experimental: added RTSP and RTP server support - added daemon mode as default (use -d for foreground mode) - added module support
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 15:32:54 UTC 2013
The following commit has been merged in the experimental branch:
commit 2effd27446662222cdb0acf2e2074247da3df177
Author: Fabrice Bellard <fabrice at bellard.org>
Date: Thu Jul 25 16:48:44 2002 +0000
added RTSP and RTP server support - added daemon mode as default (use -d for foreground mode) - added module support
Originally committed as revision 812 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/ffserver.c b/ffserver.c
index d2390f2..c7b2bc5 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -28,7 +28,6 @@
#include <errno.h>
#include <sys/time.h>
#include <time.h>
-#include <getopt.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/wait.h>
@@ -36,6 +35,9 @@
#include <netdb.h>
#include <ctype.h>
#include <signal.h>
+#include <dlfcn.h>
+
+#include "ffserver.h"
/* maximum number of simultaneous HTTP connections */
#define HTTP_MAX_CONNECTIONS 2000
@@ -44,30 +46,45 @@ enum HTTPState {
HTTPSTATE_WAIT_REQUEST,
HTTPSTATE_SEND_HEADER,
HTTPSTATE_SEND_DATA_HEADER,
- HTTPSTATE_SEND_DATA,
+ HTTPSTATE_SEND_DATA, /* sending TCP or UDP data */
HTTPSTATE_SEND_DATA_TRAILER,
- HTTPSTATE_RECEIVE_DATA,
- HTTPSTATE_WAIT_FEED,
+ 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,
};
const char *http_state[] = {
- "WAIT_REQUEST",
- "SEND_HEADER",
+ "HTTP_WAIT_REQUEST",
+ "HTTP_SEND_HEADER",
+
"SEND_DATA_HEADER",
"SEND_DATA",
"SEND_DATA_TRAILER",
"RECEIVE_DATA",
"WAIT_FEED",
+ "WAIT",
+ "WAIT_SHORT",
+ "READY",
+
+ "RTSP_WAIT_REQUEST",
+ "RTSP_SEND_REPLY",
};
#define IOBUFFER_INIT_SIZE 8192
-#define PBUFFER_INIT_SIZE 8192
/* coef for exponential mean for bitrate estimation in statistics */
#define AVG_COEF 0.9
/* timeouts are in ms */
-#define REQUEST_TIMEOUT (15 * 1000)
+#define HTTP_REQUEST_TIMEOUT (15 * 1000)
+#define RTSP_REQUEST_TIMEOUT (3600 * 24 * 1000)
+
#define SYNC_TIMEOUT (10 * 1000)
typedef struct {
@@ -91,17 +108,19 @@ typedef struct HTTPContext {
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 */
/* output format handling */
struct FFStream *stream;
/* -1 is invalid stream */
int feed_streams[MAX_STREAMS]; /* index of streams in the feed */
int switch_feed_streams[MAX_STREAMS]; /* index of streams in the feed */
int switch_pending;
- AVFormatContext fmt_ctx;
+ AVFormatContext fmt_ctx; /* instance of FFStream for one user */
int last_packet_sent; /* true if last data packet was sent */
int suppress_log;
int bandwidth;
- long start_time; /* In milliseconds - this wraps fairly often */
DataRateData datarate;
int wmp_client_id;
char protocol[16];
@@ -109,8 +128,23 @@ typedef struct HTTPContext {
char url[128];
int buffer_size;
UINT8 *buffer;
- int pbuffer_size;
- UINT8 *pbuffer;
+ int is_packetized; /* if true, the stream is packetized */
+ int packet_stream_index; /* current stream for output in state machine */
+
+ /* RTSP state specific */
+ UINT8 *pb_buffer; /* XXX: use that in all the code */
+ ByteIOContext *pb;
+ int seq; /* RTSP sequence number */
+
+ /* RTP state specific */
+ enum RTSPProtocol rtp_protocol;
+ char session_id[32]; /* session id */
+ AVFormatContext *rtp_ctx[MAX_STREAMS];
+ URLContext *rtp_handles[MAX_STREAMS];
+ /* RTP short term bandwidth limitation */
+ int packet_byte_count;
+ int packet_start_time_us; /* used for short durations (a few
+ seconds max) */
} HTTPContext;
/* each generated stream is described here */
@@ -124,7 +158,8 @@ enum StreamType {
typedef struct FFStream {
enum StreamType stream_type;
char filename[1024]; /* stream filename */
- struct FFStream *feed;
+ struct FFStream *feed; /* feed we are using (can be null if
+ coming from file) */
AVOutputFormat *fmt;
int nb_streams;
int prebuffer; /* Number of millseconds early to start */
@@ -142,8 +177,10 @@ typedef struct FFStream {
time_t pid_start; /* Of ffmpeg process */
char **child_argv;
struct FFStream *next;
+ /* RTSP options */
+ char *rtsp_option;
/* feed specific */
- int feed_opened; /* true if someone if writing to feed */
+ int feed_opened; /* true if someone is writing to the feed */
int is_feed; /* true if it is a feed */
int conns_served;
INT64 bytes_served;
@@ -158,23 +195,45 @@ typedef struct FeedData {
float avg_frame_size; /* frame size averraged over last frames with exponential mean */
} FeedData;
-struct sockaddr_in my_addr;
+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 int handle_http(HTTPContext *c);
+static void new_connection(int server_fd, int is_rtsp);
+static void close_connection(HTTPContext *c);
+
+/* HTTP handling */
+static int handle_connection(HTTPContext *c);
static int http_parse_request(HTTPContext *c);
static int http_send_data(HTTPContext *c);
static void compute_stats(HTTPContext *c);
static int open_input_stream(HTTPContext *c, const char *info);
static int http_start_receive_data(HTTPContext *c);
static int http_receive_data(HTTPContext *c);
+static int compute_send_delay(HTTPContext *c);
+
+/* RTSP handling */
+static int rtsp_parse_request(HTTPContext *c);
+static void rtsp_cmd_describe(HTTPContext *c, const char *url);
+static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPHeader *h);
+static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h);
+static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h);
+static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h);
+
+/* RTP handling */
+static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
+ FFStream *stream, const char *session_id);
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr);
static const char *my_program_name;
static int ffserver_debug;
+static int ffserver_daemon;
static int no_launch;
static int need_to_start_children;
@@ -306,13 +365,10 @@ static void start_children(FFStream *feed)
}
}
-/* main loop of the http server */
-static int http_server(struct sockaddr_in my_addr)
+/* open a listening socket */
+static int socket_open_listen(struct sockaddr_in *my_addr)
{
- int server_fd, tmp, ret;
- struct sockaddr_in from_addr;
- struct pollfd poll_table[HTTP_MAX_CONNECTIONS + 1], *poll_entry;
- HTTPContext *c, **cp;
+ int server_fd, tmp;
server_fd = socket(AF_INET,SOCK_STREAM,0);
if (server_fd < 0) {
@@ -323,7 +379,7 @@ static int http_server(struct sockaddr_in my_addr)
tmp = 1;
setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
- if (bind (server_fd, (struct sockaddr *) &my_addr, sizeof (my_addr)) < 0) {
+ if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
perror ("bind");
close(server_fd);
return -1;
@@ -334,12 +390,31 @@ static int http_server(struct sockaddr_in my_addr)
close(server_fd);
return -1;
}
+ fcntl(server_fd, F_SETFL, O_NONBLOCK);
+
+ return server_fd;
+}
+
+
+/* main loop of the http server */
+static int http_server(void)
+{
+ int server_fd, ret, rtsp_server_fd, delay, delay1;
+ struct pollfd poll_table[HTTP_MAX_CONNECTIONS + 2], *poll_entry;
+ HTTPContext *c, *c_next;
+
+ server_fd = socket_open_listen(&my_http_addr);
+ if (server_fd < 0)
+ return -1;
+ 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);
- fcntl(server_fd, F_SETFL, O_NONBLOCK);
first_http_ctx = NULL;
nb_connections = 0;
first_http_ctx = NULL;
@@ -349,40 +424,61 @@ static int http_server(struct sockaddr_in my_addr)
poll_entry->events = POLLIN;
poll_entry++;
+ poll_entry->fd = rtsp_server_fd;
+ poll_entry->events = POLLIN;
+ poll_entry++;
+
/* wait for events on each HTTP handle */
c = first_http_ctx;
+ delay = 1000;
while (c != NULL) {
int fd;
fd = c->fd;
switch(c->state) {
- case HTTPSTATE_WAIT_REQUEST:
+ case HTTPSTATE_SEND_HEADER:
+ case RTSPSTATE_SEND_REPLY:
c->poll_entry = poll_entry;
poll_entry->fd = fd;
- poll_entry->events = POLLIN;
+ poll_entry->events = POLLOUT;
poll_entry++;
break;
- case HTTPSTATE_SEND_HEADER:
case HTTPSTATE_SEND_DATA_HEADER:
case HTTPSTATE_SEND_DATA:
case HTTPSTATE_SEND_DATA_TRAILER:
- c->poll_entry = poll_entry;
- poll_entry->fd = fd;
- poll_entry->events = POLLOUT;
- poll_entry++;
+ if (!c->is_packetized) {
+ /* for TCP, we output as much as we can (may need to put a limit) */
+ c->poll_entry = poll_entry;
+ poll_entry->fd = fd;
+ poll_entry->events = POLLOUT;
+ poll_entry++;
+ } else {
+ /* not strictly correct, but currently cannot add
+ more than one fd in poll entry */
+ delay = 0;
+ }
break;
+ case HTTPSTATE_WAIT_REQUEST:
case HTTPSTATE_RECEIVE_DATA:
- c->poll_entry = poll_entry;
- poll_entry->fd = fd;
- poll_entry->events = POLLIN;
- poll_entry++;
- break;
case HTTPSTATE_WAIT_FEED:
+ case RTSPSTATE_WAIT_REQUEST:
/* need to catch errors */
c->poll_entry = poll_entry;
poll_entry->fd = fd;
poll_entry->events = POLLIN;/* Maybe this will work */
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;
@@ -393,7 +489,7 @@ static int http_server(struct sockaddr_in my_addr)
/* wait for an event on one connection. We poll at least every
second to handle timeouts */
do {
- ret = poll(poll_table, poll_entry - poll_table, 1000);
+ ret = poll(poll_table, poll_entry - poll_table, delay);
} while (ret == -1);
cur_time = gettime_ms();
@@ -404,81 +500,153 @@ static int http_server(struct sockaddr_in my_addr)
}
/* now handle the events */
-
- cp = &first_http_ctx;
- while ((*cp) != NULL) {
- c = *cp;
- if (handle_http (c) < 0) {
+ for(c = first_http_ctx; c != NULL; c = c_next) {
+ c_next = c->next;
+ if (handle_connection(c) < 0) {
/* close and free the connection */
log_connection(c);
- close(c->fd);
- if (c->fmt_in)
- av_close_input_file(c->fmt_in);
- *cp = c->next;
- nb_bandwidth -= c->bandwidth;
- av_free(c->buffer);
- av_free(c->pbuffer);
- av_free(c);
- nb_connections--;
- } else {
- cp = &c->next;
+ close_connection(c);
}
}
- /* new connection request ? */
poll_entry = poll_table;
+ /* new HTTP connection request ? */
if (poll_entry->revents & POLLIN) {
- int fd, len;
-
- len = sizeof(from_addr);
- fd = accept(server_fd, (struct sockaddr *)&from_addr,
- &len);
- if (fd >= 0) {
- fcntl(fd, F_SETFL, O_NONBLOCK);
- /* XXX: should output a warning page when coming
- close to the connection limit */
- if (nb_connections >= nb_max_connections) {
- c = NULL;
- } else {
- /* add a new connection */
- c = av_mallocz(sizeof(HTTPContext));
- if (c) {
- c->next = first_http_ctx;
- first_http_ctx = c;
- c->fd = fd;
- c->poll_entry = NULL;
- c->from_addr = from_addr;
- c->state = HTTPSTATE_WAIT_REQUEST;
- c->buffer = av_malloc(c->buffer_size = IOBUFFER_INIT_SIZE);
- c->pbuffer = av_malloc(c->pbuffer_size = PBUFFER_INIT_SIZE);
- if (!c->buffer || !c->pbuffer) {
- av_free(c->buffer);
- av_free(c->pbuffer);
- av_freep(&c);
- } else {
- c->buffer_ptr = c->buffer;
- c->buffer_end = c->buffer + c->buffer_size;
- c->timeout = cur_time + REQUEST_TIMEOUT;
- c->start_time = cur_time;
- nb_connections++;
- }
- }
- }
- if (!c) {
- close(fd);
- }
- }
+ new_connection(server_fd, 0);
}
poll_entry++;
+ /* new RTSP connection request ? */
+ if (poll_entry->revents & POLLIN) {
+ new_connection(rtsp_server_fd, 1);
+ }
}
}
-static int handle_http(HTTPContext *c)
+/* start waiting for a new HTTP/RTSP request */
+static void start_wait_request(HTTPContext *c, int is_rtsp)
{
- int len;
+ c->buffer_ptr = c->buffer;
+ c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */
+
+ if (is_rtsp) {
+ c->timeout = cur_time + RTSP_REQUEST_TIMEOUT;
+ c->state = RTSPSTATE_WAIT_REQUEST;
+ } else {
+ c->timeout = cur_time + HTTP_REQUEST_TIMEOUT;
+ c->state = HTTPSTATE_WAIT_REQUEST;
+ }
+}
+
+static void new_connection(int server_fd, int is_rtsp)
+{
+ struct sockaddr_in from_addr;
+ int fd, len;
+ HTTPContext *c = NULL;
+
+ len = sizeof(from_addr);
+ fd = accept(server_fd, (struct sockaddr *)&from_addr,
+ &len);
+ if (fd < 0)
+ return;
+ fcntl(fd, F_SETFL, O_NONBLOCK);
+
+ /* XXX: should output a warning page when coming
+ close to the connection limit */
+ if (nb_connections >= nb_max_connections)
+ goto fail;
+
+ /* add a new connection */
+ c = av_mallocz(sizeof(HTTPContext));
+ if (!c)
+ goto fail;
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
+ c->fd = fd;
+ c->poll_entry = NULL;
+ c->from_addr = from_addr;
+ c->buffer_size = IOBUFFER_INIT_SIZE;
+ c->buffer = av_malloc(c->buffer_size);
+ if (!c->buffer)
+ goto fail;
+ nb_connections++;
+
+ start_wait_request(c, is_rtsp);
+
+ return;
+
+ fail:
+ if (c) {
+ av_free(c->buffer);
+ av_free(c);
+ }
+ close(fd);
+}
+
+static void close_connection(HTTPContext *c)
+{
+ HTTPContext **cp, *c1;
+ int i, nb_streams;
+ AVFormatContext *ctx;
+ URLContext *h;
+ AVStream *st;
+
+ /* remove connection from list */
+ cp = &first_http_ctx;
+ while ((*cp) != NULL) {
+ c1 = *cp;
+ if (c1 == c) {
+ *cp = c->next;
+ } else {
+ cp = &c1->next;
+ }
+ }
+
+ /* remove connection associated resources */
+ if (c->fd >= 0)
+ close(c->fd);
+ if (c->fmt_in) {
+ /* close each frame parser */
+ for(i=0;i<c->fmt_in->nb_streams;i++) {
+ st = c->fmt_in->streams[i];
+ if (st->codec.codec) {
+ avcodec_close(&st->codec);
+ }
+ }
+ av_close_input_file(c->fmt_in);
+ }
+
+ /* free RTP output streams if any */
+ nb_streams = 0;
+ if (c->stream)
+ nb_streams = c->stream->nb_streams;
+
+ for(i=0;i<nb_streams;i++) {
+ ctx = c->rtp_ctx[i];
+ if (ctx) {
+ av_write_trailer(ctx);
+ av_free(ctx);
+ }
+ h = c->rtp_handles[i];
+ if (h) {
+ url_close(h);
+ }
+ }
+
+ nb_bandwidth -= c->bandwidth;
+ av_freep(&c->pb_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:
/* timeout ? */
if ((c->timeout - cur_time) < 0)
return -1;
@@ -503,7 +671,12 @@ static int handle_http(HTTPContext *c)
if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
(ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
/* request found : parse it and reply */
- if (http_parse_request(c) < 0)
+ if (c->state == HTTPSTATE_WAIT_REQUEST) {
+ ret = http_parse_request(c);
+ } else {
+ ret = rtsp_parse_request(c);
+ }
+ if (ret < 0)
return -1;
} else if (ptr >= c->buffer_end) {
/* request too long: cannot do anything */
@@ -516,13 +689,14 @@ static int handle_http(HTTPContext *c)
if (c->poll_entry->revents & (POLLERR | POLLHUP))
return -1;
- /* no need to read if no events */
+ /* no need to write if no events */
if (!(c->poll_entry->revents & POLLOUT))
return 0;
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
if (len < 0) {
if (errno != EAGAIN && errno != EINTR) {
/* error : close connection */
+ av_freep(&c->pb_buffer);
return -1;
}
} else {
@@ -531,10 +705,12 @@ static int handle_http(HTTPContext *c)
c->stream->bytes_served += len;
c->data_count += len;
if (c->buffer_ptr >= c->buffer_end) {
+ av_freep(&c->pb_buffer);
/* if error, exit */
- if (c->http_error)
+ if (c->http_error) {
return -1;
- /* all the buffer was send : synchronize to the incoming stream */
+ }
+ /* all the buffer was sent : synchronize to the incoming stream */
c->state = HTTPSTATE_SEND_DATA_HEADER;
c->buffer_ptr = c->buffer_end = c->buffer;
}
@@ -544,12 +720,17 @@ static int handle_http(HTTPContext *c)
case HTTPSTATE_SEND_DATA:
case HTTPSTATE_SEND_DATA_HEADER:
case HTTPSTATE_SEND_DATA_TRAILER:
- /* no need to read if no events */
- if (c->poll_entry->revents & (POLLERR | POLLHUP))
- return -1;
-
- if (!(c->poll_entry->revents & POLLOUT))
- return 0;
+ /* for packetized output, we consider we can always write (the
+ input streams sets the speed). It may be better to verify
+ that we do not rely too much on the kernel queues */
+ if (!c->is_packetized) {
+ if (c->poll_entry->revents & (POLLERR | POLLHUP))
+ return -1;
+
+ /* no need to read if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ }
if (http_send_data(c) < 0)
return -1;
break;
@@ -569,6 +750,45 @@ static int handle_http(HTTPContext *c)
/* 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);
+ return -1;
+ }
+ /* no need to write if no events */
+ if (!(c->poll_entry->revents & POLLOUT))
+ return 0;
+ len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
+ if (len < 0) {
+ if (errno != EAGAIN && errno != EINTR) {
+ /* error : close connection */
+ av_freep(&c->pb_buffer);
+ return -1;
+ }
+ } else {
+ c->buffer_ptr += len;
+ c->data_count += len;
+ if (c->buffer_ptr >= c->buffer_end) {
+ /* all the buffer was sent : wait for a new request */
+ av_freep(&c->pb_buffer);
+ start_wait_request(c, 1);
+ }
+ }
+ break;
+ case HTTPSTATE_READY:
+ /* nothing to do */
+ break;
default:
return -1;
}
@@ -708,6 +928,35 @@ static void do_switch_stream(HTTPContext *c, int i)
c->switch_feed_streams[i] = -1;
}
+/* XXX: factorize in utils.c ? */
+/* XXX: take care with different space meaning */
+static void skip_spaces(const char **pp)
+{
+ const char *p;
+ p = *pp;
+ while (*p == ' ' || *p == '\t')
+ p++;
+ *pp = p;
+}
+
+static void get_word(char *buf, int buf_size, const char **pp)
+{
+ const char *p;
+ char *q;
+
+ p = *pp;
+ skip_spaces(&p);
+ q = buf;
+ while (!isspace(*p) && *p != '\0') {
+ if ((q - buf) < buf_size - 1)
+ *q++ = *p;
+ p++;
+ }
+ if (buf_size > 0)
+ *q = '\0';
+ *pp = p;
+}
+
/* parse http request and prepare header */
static int http_parse_request(HTTPContext *c)
{
@@ -716,6 +965,7 @@ static int http_parse_request(HTTPContext *c)
int doing_asx;
int doing_asf_redirector;
int doing_ram;
+ int doing_rtsp_redirector;
char cmd[32];
char info[1024], *filename;
char url[1024], *q;
@@ -728,14 +978,7 @@ static int http_parse_request(HTTPContext *c)
char *useragent = 0;
p = c->buffer;
- q = cmd;
- while (!isspace(*p) && *p != '\0') {
- if ((q - cmd) < sizeof(cmd) - 1)
- *q++ = *p;
- p++;
- }
- *q = '\0';
-
+ get_word(cmd, sizeof(cmd), (const char **)&p);
pstrcpy(c->method, sizeof(c->method), cmd);
if (!strcmp(cmd, "GET"))
@@ -745,25 +988,10 @@ static int http_parse_request(HTTPContext *c)
else
return -1;
- while (isspace(*p)) p++;
- q = url;
- while (!isspace(*p) && *p != '\0') {
- if ((q - url) < sizeof(url) - 1)
- *q++ = *p;
- p++;
- }
- *q = '\0';
-
+ get_word(url, sizeof(url), (const char **)&p);
pstrcpy(c->url, sizeof(c->url), url);
- while (isspace(*p)) p++;
- q = protocol;
- while (!isspace(*p) && *p != '\0') {
- if ((q - protocol) < sizeof(protocol) - 1)
- *q++ = *p;
- p++;
- }
- *q = '\0';
+ get_word(protocol, sizeof(protocol), (const char **)&p);
if (strcmp(protocol, "HTTP/1.0") && strcmp(protocol, "HTTP/1.1"))
return -1;
@@ -820,6 +1048,33 @@ static int http_parse_request(HTTPContext *c)
doing_ram = 0;
}
+ if (strlen(filename) > 5 &&
+ strcmp(".rtsp", filename + strlen(filename) - 5) == 0) {
+ char file1[1024];
+ char file2[1024];
+ char *p;
+
+ doing_rtsp_redirector = 1;
+ /* compute filename by matching without the file extensions */
+ pstrcpy(file1, sizeof(file1), filename);
+ p = strrchr(file1, '.');
+ if (p)
+ *p = '\0';
+ for(stream = first_stream; stream != NULL; stream = stream->next) {
+ pstrcpy(file2, sizeof(file2), stream->filename);
+ p = strrchr(file2, '.');
+ if (p)
+ *p = '\0';
+ if (!strcmp(file1, file2)) {
+ pstrcpy(url, sizeof(url), stream->filename);
+ filename = url;
+ break;
+ }
+ }
+ } else {
+ doing_rtsp_redirector = 0;
+ }
+
stream = first_stream;
while (stream != NULL) {
if (!strcmp(stream->filename, filename))
@@ -902,7 +1157,8 @@ static int http_parse_request(HTTPContext *c)
return 0;
}
- if (doing_asx || doing_ram || doing_asf_redirector) {
+ if (doing_asx || doing_ram || doing_asf_redirector ||
+ doing_rtsp_redirector) {
char *hostinfo = 0;
for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
@@ -958,8 +1214,23 @@ static int http_parse_request(HTTPContext *c)
q += sprintf(q, "[Reference]\r\n");
q += sprintf(q, "Ref1=http://%s/%s%s\r\n",
hostbuf, filename, info);
- } else
+ } else if (doing_rtsp_redirector) {
+ char hostname[256], *p;
+ /* extract only hostname */
+ pstrcpy(hostname, sizeof(hostname), hostbuf);
+ p = strrchr(hostname, ':');
+ if (p)
+ *p = '\0';
+ q += sprintf(q, "HTTP/1.0 200 RTSP Redirect follows\r\n");
+ /* XXX: incorrect mime type ? */
+ q += sprintf(q, "Content-type: application/x-rtsp\r\n");
+ q += sprintf(q, "\r\n");
+ q += sprintf(q, "rtsp://%s:%d/%s\r\n",
+ hostname, ntohs(my_rtsp_addr.sin_port),
+ filename);
+ } else {
av_abort();
+ }
/* prepare output buffer */
c->buffer_ptr = c->buffer;
@@ -1116,7 +1387,7 @@ static int http_parse_request(HTTPContext *c)
return 0;
}
-static int fmt_bytecount(char *q, INT64 count)
+static void fmt_bytecount(ByteIOContext *pb, INT64 count)
{
static const char *suffix = " kMGTP";
const char *s;
@@ -1124,64 +1395,68 @@ static int fmt_bytecount(char *q, INT64 count)
for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++) {
}
- return sprintf(q, "%lld%c", count, *s);
+ url_fprintf(pb, "%lld%c", count, *s);
}
static void compute_stats(HTTPContext *c)
{
HTTPContext *c1;
FFStream *stream;
- char *q, *p;
+ char *p;
time_t ti;
- int i;
- char *new_buffer;
+ int i, len;
+ ByteIOContext pb1, *pb = &pb1;
- new_buffer = av_malloc(65536);
- if (new_buffer) {
- av_free(c->buffer);
- c->buffer_size = 65536;
- c->buffer = new_buffer;
+ if (url_open_dyn_buf(pb) < 0) {
+ /* XXX: return an error ? */
c->buffer_ptr = c->buffer;
- c->buffer_end = c->buffer + c->buffer_size;
+ c->buffer_end = c->buffer;
+ return;
}
- q = c->buffer;
- q += sprintf(q, "HTTP/1.0 200 OK\r\n");
- q += sprintf(q, "Content-type: %s\r\n", "text/html");
- q += sprintf(q, "Pragma: no-cache\r\n");
- q += sprintf(q, "\r\n");
+ url_fprintf(pb, "HTTP/1.0 200 OK\r\n");
+ url_fprintf(pb, "Content-type: %s\r\n", "text/html");
+ url_fprintf(pb, "Pragma: no-cache\r\n");
+ url_fprintf(pb, "\r\n");
- q += sprintf(q, "<HEAD><TITLE>FFServer Status</TITLE>\n");
+ url_fprintf(pb, "<HEAD><TITLE>FFServer Status</TITLE>\n");
if (c->stream->feed_filename) {
- q += sprintf(q, "<link rel=\"shortcut icon\" href=\"%s\">\n", c->stream->feed_filename);
+ url_fprintf(pb, "<link rel=\"shortcut icon\" href=\"%s\">\n", c->stream->feed_filename);
}
- q += sprintf(q, "</HEAD>\n<BODY>");
- q += sprintf(q, "<H1>FFServer Status</H1>\n");
+ url_fprintf(pb, "</HEAD>\n<BODY>");
+ url_fprintf(pb, "<H1>FFServer Status</H1>\n");
/* format status */
- q += sprintf(q, "<H2>Available Streams</H2>\n");
- q += sprintf(q, "<TABLE cellspacing=0 cellpadding=4>\n");
- q += sprintf(q, "<TR><Th valign=top>Path<th align=left>Served<br>Conns<Th><br>bytes<Th valign=top>Format<Th>Bit rate<br>kbits/s<Th align=left>Video<br>kbits/s<th><br>Codec<Th align=left>Audio<br>kbits/s<th><br>Codec<Th align=left valign=top>Feed\n");
+ url_fprintf(pb, "<H2>Available Streams</H2>\n");
+ url_fprintf(pb, "<TABLE cellspacing=0 cellpadding=4>\n");
+ url_fprintf(pb, "<TR><Th valign=top>Path<th align=left>Served<br>Conns<Th><br>bytes<Th valign=top>Format<Th>Bit rate<br>kbits/s<Th align=left>Video<br>kbits/s<th><br>Codec<Th align=left>Audio<br>kbits/s<th><br>Codec<Th align=left valign=top>Feed\n");
stream = first_stream;
while (stream != NULL) {
char sfilename[1024];
char *eosf;
if (stream->feed != stream) {
- pstrcpy(sfilename, sizeof(sfilename) - 1, stream->filename);
+ pstrcpy(sfilename, sizeof(sfilename) - 10, stream->filename);
eosf = sfilename + strlen(sfilename);
if (eosf - sfilename >= 4) {
if (strcmp(eosf - 4, ".asf") == 0) {
strcpy(eosf - 4, ".asx");
} else if (strcmp(eosf - 3, ".rm") == 0) {
strcpy(eosf - 3, ".ram");
+ } else if (stream->fmt == &rtp_mux) {
+ /* generate a sample RTSP director - maybe should
+ generate a .sdp file ? */
+ eosf = strrchr(sfilename, '.');
+ if (!eosf)
+ eosf = sfilename + strlen(sfilename);
+ strcpy(eosf, ".rtsp");
}
}
- q += sprintf(q, "<TR><TD><A HREF=\"/%s\">%s</A> ",
+ url_fprintf(pb, "<TR><TD><A HREF=\"/%s\">%s</A> ",
sfilename, stream->filename);
- q += sprintf(q, "<td align=right> %d <td align=right> ",
+ url_fprintf(pb, "<td align=right> %d <td align=right> ",
stream->conns_served);
- q += fmt_bytecount(q, stream->bytes_served);
+ fmt_bytecount(pb, stream->bytes_served);
switch(stream->stream_type) {
case STREAM_TYPE_LIVE:
{
@@ -1216,58 +1491,63 @@ static void compute_stats(HTTPContext *c)
av_abort();
}
}
- q += sprintf(q, "<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,
(audio_bit_rate + video_bit_rate) / 1000,
video_bit_rate / 1000, video_codec_name, video_codec_name_extra,
audio_bit_rate / 1000, audio_codec_name, audio_codec_name_extra);
if (stream->feed) {
- q += sprintf(q, "<TD>%s", stream->feed->filename);
+ url_fprintf(pb, "<TD>%s", stream->feed->filename);
} else {
- q += sprintf(q, "<TD>%s", stream->feed_filename);
+ url_fprintf(pb, "<TD>%s", stream->feed_filename);
}
- q += sprintf(q, "\n");
+ url_fprintf(pb, "\n");
}
break;
default:
- q += sprintf(q, "<TD align=center> - <TD align=right> - <TD align=right> - <td><td align=right> - <TD>\n");
+ url_fprintf(pb, "<TD align=center> - <TD align=right> - <TD align=right> - <td><td align=right> - <TD>\n");
break;
}
}
stream = stream->next;
}
- q += sprintf(q, "</TABLE>\n");
+ url_fprintf(pb, "</TABLE>\n");
stream = first_stream;
while (stream != NULL) {
if (stream->feed == stream) {
- q += sprintf(q, "<h2>Feed %s</h2>", stream->filename);
+ url_fprintf(pb, "<h2>Feed %s</h2>", stream->filename);
if (stream->pid) {
- FILE *pid_stat;
- char ps_cmd[64];
-
- q += sprintf(q, "Running as pid %d.\n", stream->pid);
-
-#ifdef linux
- /* This is somewhat linux specific I guess */
- snprintf(ps_cmd, sizeof(ps_cmd), "ps -o \"%%cpu,bsdtime\" --no-headers %d", stream->pid);
-
- pid_stat = popen(ps_cmd, "r");
- if (pid_stat) {
- char cpuperc[10];
- char cpuused[64];
+ url_fprintf(pb, "Running as pid %d.\n", stream->pid);
- if (fscanf(pid_stat, "%10s %64s", cpuperc, cpuused) == 2) {
- q += sprintf(q, "Currently using %s%% of the cpu. Total time used %s.\n",
- cpuperc, cpuused);
+#if defined(linux) && !defined(CONFIG_NOCUTILS)
+ {
+ FILE *pid_stat;
+ char ps_cmd[64];
+
+ /* This is somewhat linux specific I guess */
+ snprintf(ps_cmd, sizeof(ps_cmd),
+ "ps -o \"%%cpu,cputime\" --no-headers %d",
+ stream->pid);
+
+ pid_stat = popen(ps_cmd, "r");
+ if (pid_stat) {
+ char cpuperc[10];
+ char cpuused[64];
+
+ if (fscanf(pid_stat, "%10s %64s", cpuperc,
+ cpuused) == 2) {
+ url_fprintf(pb, "Currently using %s%% of the cpu. Total time used %s.\n",
+ cpuperc, cpuused);
+ }
+ fclose(pid_stat);
}
- fclose(pid_stat);
}
#endif
- q += sprintf(q, "<p>");
+ url_fprintf(pb, "<p>");
}
- q += sprintf(q, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>type<th>kbits/s<th align=left>codec<th align=left>Parameters\n");
+ url_fprintf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>type<th>kbits/s<th align=left>codec<th align=left>Parameters\n");
for (i = 0; i < stream->nb_streams; i++) {
AVStream *st = stream->streams[i];
@@ -1289,10 +1569,10 @@ static void compute_stats(HTTPContext *c)
default:
av_abort();
}
- q += sprintf(q, "<tr><td align=right>%d<td>%s<td align=right>%d<td>%s<td>%s\n",
+ 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);
}
- q += sprintf(q, "</table>\n");
+ url_fprintf(pb, "</table>\n");
}
stream = stream->next;
@@ -1307,9 +1587,9 @@ static void compute_stats(HTTPContext *c)
/* feed status */
stream = first_feed;
while (stream != NULL) {
- q += sprintf(q, "<H1>Feed '%s'</H1>\n", stream->filename);
- q += sprintf(q, "<TABLE>\n");
- q += sprintf(q, "<TR><TD>Parameters<TD>Frame count<TD>Size<TD>Avg bitrate (kbits/s)\n");
+ url_fprintf(pb, "<H1>Feed '%s'</H1>\n", stream->filename);
+ url_fprintf(pb, "<TABLE>\n");
+ url_fprintf(pb, "<TR><TD>Parameters<TD>Frame count<TD>Size<TD>Avg bitrate (kbits/s)\n");
for(i=0;i<stream->nb_streams;i++) {
AVStream *st = stream->streams[i];
FeedData *fdata = st->priv_data;
@@ -1319,98 +1599,90 @@ static void compute_stats(HTTPContext *c)
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;
- q += sprintf(q, "<TR><TD>%s <TD> %d <TD> %Ld <TD> %0.1f\n",
+ url_fprintf(pb, "<TR><TD>%s <TD> %d <TD> %Ld <TD> %0.1f\n",
buf, enc->frame_number, fdata->data_count, avg / 1000.0);
}
- q += sprintf(q, "</TABLE>\n");
+ url_fprintf(pb, "</TABLE>\n");
stream = stream->next_feed;
}
}
#endif
/* connection status */
- q += sprintf(q, "<H2>Connection Status</H2>\n");
+ url_fprintf(pb, "<H2>Connection Status</H2>\n");
- q += sprintf(q, "Number of connections: %d / %d<BR>\n",
+ url_fprintf(pb, "Number of connections: %d / %d<BR>\n",
nb_connections, nb_max_connections);
- q += sprintf(q, "Bandwidth in use: %dk / %dk<BR>\n",
+ url_fprintf(pb, "Bandwidth in use: %dk / %dk<BR>\n",
nb_bandwidth, nb_max_bandwidth);
- q += sprintf(q, "<TABLE>\n");
- q += sprintf(q, "<TR><Th>#<Th>File<Th>IP<Th>State<Th>Target bits/sec<Th>Actual bits/sec<Th>Bytes transferred\n");
+ url_fprintf(pb, "<TABLE>\n");
+ url_fprintf(pb, "<TR><th>#<th>File<th>IP<th>Proto<th>State<th>Target bits/sec<th>Actual bits/sec<th>Bytes transferred\n");
c1 = first_http_ctx;
i = 0;
- while (c1 != NULL && q < (char *) c->buffer + c->buffer_size - 2048) {
+ while (c1 != NULL) {
int bitrate;
int j;
bitrate = 0;
- for (j = 0; j < c1->stream->nb_streams; j++) {
- if (c1->feed_streams[j] >= 0) {
- bitrate += c1->stream->feed->streams[c1->feed_streams[j]]->codec.bit_rate;
+ if (c1->stream) {
+ for (j = 0; j < c1->stream->nb_streams; j++) {
+ if (!c1->stream->feed) {
+ bitrate += c1->stream->streams[j]->codec.bit_rate;
+ } else {
+ if (c1->feed_streams[j] >= 0) {
+ bitrate += c1->stream->feed->streams[c1->feed_streams[j]]->codec.bit_rate;
+ }
+ }
}
}
i++;
p = inet_ntoa(c1->from_addr.sin_addr);
- q += sprintf(q, "<TR><TD><B>%d</B><TD>%s%s <TD> %s <TD> %s <td align=right>",
- i, c1->stream->filename,
- c1->state == HTTPSTATE_RECEIVE_DATA ? "(input)" : "",
- p,
- http_state[c1->state]);
- q += fmt_bytecount(q, bitrate);
- q += sprintf(q, "<td align=right>");
- q += fmt_bytecount(q, compute_datarate(&c1->datarate, c1->data_count) * 8);
- q += sprintf(q, "<td align=right>");
- q += fmt_bytecount(q, c1->data_count);
- *q++ = '\n';
+ url_fprintf(pb, "<TR><TD><B>%d</B><TD>%s%s<TD>%s<TD>%s<TD>%s<td align=right>",
+ i,
+ c1->stream ? c1->stream->filename : "",
+ c1->state == HTTPSTATE_RECEIVE_DATA ? "(input)" : "",
+ p,
+ c1->protocol,
+ http_state[c1->state]);
+ fmt_bytecount(pb, bitrate);
+ url_fprintf(pb, "<td align=right>");
+ fmt_bytecount(pb, compute_datarate(&c1->datarate, c1->data_count) * 8);
+ url_fprintf(pb, "<td align=right>");
+ fmt_bytecount(pb, c1->data_count);
+ url_fprintf(pb, "\n");
c1 = c1->next;
}
- q += sprintf(q, "</TABLE>\n");
+ url_fprintf(pb, "</TABLE>\n");
/* date */
ti = time(NULL);
p = ctime(&ti);
- q += sprintf(q, "<HR size=1 noshade>Generated at %s", p);
- q += sprintf(q, "</BODY>\n</HTML>\n");
+ url_fprintf(pb, "<HR size=1 noshade>Generated at %s", p);
+ url_fprintf(pb, "</BODY>\n</HTML>\n");
- c->buffer_ptr = c->buffer;
- c->buffer_end = q;
+ len = url_close_dyn_buf(pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
}
-
-static void http_write_packet(void *opaque,
- unsigned char *buf, int size)
+/* check if the parser needs to be opened for stream i */
+static void open_parser(AVFormatContext *s, int i)
{
- HTTPContext *c = opaque;
-
- if (c->buffer_ptr == c->buffer_end || !c->buffer_ptr)
- c->buffer_ptr = c->buffer_end = c->buffer;
-
- if (c->buffer_end - c->buffer + size > c->buffer_size) {
- int new_buffer_size = c->buffer_size * 2;
- UINT8 *new_buffer;
-
- if (new_buffer_size <= c->buffer_end - c->buffer + size) {
- new_buffer_size = c->buffer_end - c->buffer + size + c->buffer_size;
- }
+ AVStream *st = s->streams[i];
+ AVCodec *codec;
- new_buffer = av_malloc(new_buffer_size);
- if (new_buffer) {
- memcpy(new_buffer, c->buffer, c->buffer_end - c->buffer);
- c->buffer_end += (new_buffer - c->buffer);
- c->buffer_ptr += (new_buffer - c->buffer);
- av_free(c->buffer);
- c->buffer = new_buffer;
- c->buffer_size = new_buffer_size;
- } else {
- av_abort();
+ 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;
+ }
}
}
-
- memcpy(c->buffer_end, buf, size);
- c->buffer_end += size;
}
static int open_input_stream(HTTPContext *c, const char *info)
@@ -1418,7 +1690,7 @@ static int open_input_stream(HTTPContext *c, const char *info)
char buf[128];
char input_filename[1024];
AVFormatContext *s;
- int buf_size;
+ int buf_size, i;
INT64 stream_pos;
/* find file name */
@@ -1448,131 +1720,253 @@ static int open_input_stream(HTTPContext *c, const char *info)
return -1;
/* open stream */
- if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 0)
+ if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 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);
+
+ /* choose stream as clock source (we favorize video stream if
+ present) for packet sending */
+ c->pts_stream_index = 0;
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->pts_stream_index == 0 &&
+ c->stream->streams[i]->codec.codec_type == CODEC_TYPE_VIDEO) {
+ c->pts_stream_index = i;
+ }
+ }
if (c->fmt_in->iformat->read_seek) {
c->fmt_in->iformat->read_seek(c->fmt_in, stream_pos);
}
-
- // printf("stream %s opened pos=%0.6f\n", input_filename, stream_pos / 1000000.0);
+ /* set the start time (needed for maxtime and RTP packet timing) */
+ c->start_time = cur_time;
+ c->first_pts = AV_NOPTS_VALUE;
+ printf("stream %s opened pos=%0.6f\n", input_filename, stream_pos / 1000000.0);
return 0;
}
-static int http_prepare_data(HTTPContext *c)
-{
- int i;
+/* currently desactivated because the new PTS handling is not
+ satisfactory yet */
+//#define AV_READ_FRAME
+#ifdef AV_READ_FRAME
- switch(c->state) {
- case HTTPSTATE_SEND_DATA_HEADER:
- memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
- pstrcpy(c->fmt_ctx.author, sizeof(c->fmt_ctx.author), c->stream->author);
- pstrcpy(c->fmt_ctx.comment, sizeof(c->fmt_ctx.comment), c->stream->comment);
- pstrcpy(c->fmt_ctx.copyright, sizeof(c->fmt_ctx.copyright), c->stream->copyright);
- pstrcpy(c->fmt_ctx.title, sizeof(c->fmt_ctx.title), c->stream->title);
-
- if (c->stream->feed) {
- /* open output stream by using specified codecs */
- c->fmt_ctx.oformat = c->stream->fmt;
- c->fmt_ctx.nb_streams = c->stream->nb_streams;
- for(i=0;i<c->fmt_ctx.nb_streams;i++) {
- AVStream *st;
- st = av_mallocz(sizeof(AVStream));
- c->fmt_ctx.streams[i] = st;
- if (c->stream->feed == c->stream)
- memcpy(st, c->stream->streams[i], sizeof(AVStream));
- else
- memcpy(st, c->stream->feed->streams[c->stream->feed_streams[i]], sizeof(AVStream));
+/* 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;
- st->codec.frame_number = 0; /* XXX: should be done in
- AVStream, not in codec */
+ /* 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.key_frame)
+ pkt->flags |= PKT_FLAG_KEY;
+ return 0;
+ }
}
- c->got_key_frame = 0;
} else {
- /* open output stream by using codecs in specified file */
- c->fmt_ctx.oformat = c->stream->fmt;
- c->fmt_ctx.nb_streams = c->fmt_in->nb_streams;
- for(i=0;i<c->fmt_ctx.nb_streams;i++) {
- AVStream *st;
- st = av_mallocz(sizeof(AVStream));
- c->fmt_ctx.streams[i] = st;
- memcpy(st, c->fmt_in->streams[i], sizeof(AVStream));
- st->codec.frame_number = 0; /* XXX: should be done in
- AVStream, not in codec */
+ /* 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;
}
- c->got_key_frame = 0;
}
- init_put_byte(&c->fmt_ctx.pb, c->pbuffer, c->pbuffer_size,
- 1, c, NULL, http_write_packet, NULL);
- c->fmt_ctx.pb.is_streamed = 1;
- /* prepare header */
- av_write_header(&c->fmt_ctx);
- c->state = HTTPSTATE_SEND_DATA;
+ }
+}
+
+static int compute_send_delay(HTTPContext *c)
+{
+ INT64 cur_pts, delta_pts, next_pts;
+ int delay1;
+
+ /* compute current pts value from system time */
+ cur_pts = ((INT64)(cur_time - c->start_time) * c->fmt_in->pts_den) /
+ (c->fmt_in->pts_num * 1000LL);
+ /* compute the delta from the stream we choose as
+ main clock (we do that to avoid using explicit
+ buffers to do exact packet reordering for each
+ stream */
+ /* XXX: really need to fix the number of streams */
+ if (c->pts_stream_index >= c->fmt_in->nb_streams)
+ next_pts = cur_pts;
+ else
+ next_pts = c->fmt_in->streams[c->pts_stream_index]->pts.val;
+ delta_pts = next_pts - cur_pts;
+ if (delta_pts <= 0) {
+ delay1 = 0;
+ } else {
+ delay1 = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
+ }
+ return delay1;
+}
+#else
+
+/* just fall backs */
+int av_read_frame(AVFormatContext *s, AVPacket *pkt)
+{
+ return av_read_packet(s, pkt);
+}
+
+static int compute_send_delay(HTTPContext *c)
+{
+ return 0;
+}
+
+#endif
+
+static int http_prepare_data(HTTPContext *c)
+{
+ int i, len, ret;
+ AVFormatContext *ctx;
+
+ switch(c->state) {
+ case HTTPSTATE_SEND_DATA_HEADER:
+ memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
+ pstrcpy(c->fmt_ctx.author, sizeof(c->fmt_ctx.author),
+ c->stream->author);
+ pstrcpy(c->fmt_ctx.comment, sizeof(c->fmt_ctx.comment),
+ c->stream->comment);
+ pstrcpy(c->fmt_ctx.copyright, sizeof(c->fmt_ctx.copyright),
+ c->stream->copyright);
+ pstrcpy(c->fmt_ctx.title, sizeof(c->fmt_ctx.title),
+ c->stream->title);
+
+ /* open output stream by using specified codecs */
+ c->fmt_ctx.oformat = c->stream->fmt;
+ c->fmt_ctx.nb_streams = c->stream->nb_streams;
+ for(i=0;i<c->fmt_ctx.nb_streams;i++) {
+ AVStream *st;
+ st = av_mallocz(sizeof(AVStream));
+ c->fmt_ctx.streams[i] = st;
+ /* if file or feed, then just take streams from FFStream struct */
+ if (!c->stream->feed ||
+ c->stream->feed == c->stream)
+ memcpy(st, c->stream->streams[i], sizeof(AVStream));
+ else
+ memcpy(st, c->stream->feed->streams[c->stream->feed_streams[i]],
+ sizeof(AVStream));
+ st->codec.frame_number = 0; /* XXX: should be done in
+ AVStream, not in codec */
+ }
+ c->got_key_frame = 0;
+
+ /* prepare header and save header data in a stream */
+ if (url_open_dyn_buf(&c->fmt_ctx.pb) < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ c->fmt_ctx.pb.is_streamed = 1;
+
+ av_write_header(&c->fmt_ctx);
+
+ len = url_close_dyn_buf(&c->fmt_ctx.pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
+ c->state = HTTPSTATE_SEND_DATA;
c->last_packet_sent = 0;
break;
case HTTPSTATE_SEND_DATA:
/* find a new packet */
-#if 0
- fifo_total_size = http_fifo_write_count - c->last_http_fifo_write_count;
- if (fifo_total_size >= ((3 * FIFO_MAX_SIZE) / 4)) {
- /* overflow : resync. We suppose that wptr is at this
- point a pointer to a valid packet */
- c->rptr = http_fifo.wptr;
- c->got_key_frame = 0;
- }
-
- start_rptr = c->rptr;
- if (fifo_read(&http_fifo, (UINT8 *)&hdr, sizeof(hdr), &c->rptr) < 0)
- return 0;
- payload_size = ntohs(hdr.payload_size);
- payload = av_malloc(payload_size);
- if (fifo_read(&http_fifo, payload, payload_size, &c->rptr) < 0) {
- /* cannot read all the payload */
- av_free(payload);
- c->rptr = start_rptr;
- return 0;
- }
-
- c->last_http_fifo_write_count = http_fifo_write_count -
- fifo_size(&http_fifo, c->rptr);
-
- if (c->stream->stream_type != STREAM_TYPE_MASTER) {
- /* test if the packet can be handled by this format */
- ret = 0;
- for(i=0;i<c->fmt_ctx.nb_streams;i++) {
- AVStream *st = c->fmt_ctx.streams[i];
- if (test_header(&hdr, &st->codec)) {
- /* only begin sending when got a key frame */
- if (st->codec.key_frame)
- c->got_key_frame |= 1 << i;
- if (c->got_key_frame & (1 << i)) {
- ret = c->fmt_ctx.format->write_packet(&c->fmt_ctx, i,
- payload, payload_size);
- }
- break;
- }
- }
- if (ret) {
- /* must send trailer now */
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
- }
- } else {
- /* master case : send everything */
- char *q;
- q = c->buffer;
- memcpy(q, &hdr, sizeof(hdr));
- q += sizeof(hdr);
- memcpy(q, payload, payload_size);
- q += payload_size;
- c->buffer_ptr = c->buffer;
- c->buffer_end = q;
- }
- av_free(payload);
-#endif
{
AVPacket pkt;
-
+
/* read a packet from the input stream */
if (c->stream->feed) {
ffm_set_write_index(c->fmt_in,
@@ -1584,95 +1978,150 @@ static int http_prepare_data(HTTPContext *c)
c->stream->max_time + c->start_time - cur_time < 0) {
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
- } else if (av_read_packet(c->fmt_in, &pkt) < 0) {
- if (c->stream->feed && c->stream->feed->feed_opened) {
- /* if coming from feed, it means we reached the end of the
- ffm file, so must wait for more data */
- c->state = HTTPSTATE_WAIT_FEED;
- return 1; /* state changed */
- } else {
- /* must send trailer now because eof or error */
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
- }
} else {
- /* send it to the appropriate stream */
- if (c->stream->feed) {
- /* if coming from a feed, select the right stream */
- if (c->switch_pending) {
- c->switch_pending = 0;
+ if (c->is_packetized) {
+ if (compute_send_delay(c) > 0) {
+ c->state = HTTPSTATE_WAIT;
+ return 1; /* state changed */
+ }
+ }
+ if (av_read_frame(c->fmt_in, &pkt) < 0) {
+ if (c->stream->feed && c->stream->feed->feed_opened) {
+ /* if coming from feed, it means we reached the end of the
+ ffm file, so must wait for more data */
+ c->state = HTTPSTATE_WAIT_FEED;
+ return 1; /* state changed */
+ } else {
+ /* must send trailer now because eof or error */
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
+ } else {
+ /* update first pts if needed */
+ if (c->first_pts == AV_NOPTS_VALUE)
+ c->first_pts = pkt.pts;
+
+ /* send it to the appropriate stream */
+ if (c->stream->feed) {
+ /* if coming from a feed, select the right stream */
+ if (c->switch_pending) {
+ c->switch_pending = 0;
+ for(i=0;i<c->stream->nb_streams;i++) {
+ if (c->switch_feed_streams[i] == pkt.stream_index) {
+ if (pkt.flags & PKT_FLAG_KEY) {
+ do_switch_stream(c, i);
+ }
+ }
+ if (c->switch_feed_streams[i] >= 0) {
+ c->switch_pending = 1;
+ }
+ }
+ }
for(i=0;i<c->stream->nb_streams;i++) {
- if (c->switch_feed_streams[i] == pkt.stream_index) {
+ if (c->feed_streams[i] == pkt.stream_index) {
+ pkt.stream_index = i;
if (pkt.flags & PKT_FLAG_KEY) {
- do_switch_stream(c, i);
+ c->got_key_frame |= 1 << i;
+ }
+ /* See if we have all the key frames, then
+ * we start to send. This logic is not quite
+ * right, but it works for the case of a
+ * single video stream with one or more
+ * audio streams (for which every frame is
+ * typically a key frame).
+ */
+ if (!c->stream->send_on_key ||
+ ((c->got_key_frame + 1) >> c->stream->nb_streams)) {
+ goto send_it;
}
- }
- if (c->switch_feed_streams[i] >= 0) {
- c->switch_pending = 1;
}
}
- }
- for(i=0;i<c->stream->nb_streams;i++) {
- if (c->feed_streams[i] == pkt.stream_index) {
- pkt.stream_index = i;
- if (pkt.flags & PKT_FLAG_KEY) {
- c->got_key_frame |= 1 << i;
- }
- /* See if we have all the key frames, then
- * we start to send. This logic is not quite
- * right, but it works for the case of a
- * single video stream with one or more
- * audio streams (for which every frame is
- * typically a key frame).
- */
- if (!c->stream->send_on_key || ((c->got_key_frame + 1) >> c->stream->nb_streams)) {
- goto send_it;
- }
+ } 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) {
+ c->packet_stream_index = pkt.stream_index;
+ ctx = c->rtp_ctx[c->packet_stream_index];
+ codec = &ctx->streams[0]->codec;
+ } else {
+ ctx = &c->fmt_ctx;
+ /* Fudge here */
+ codec = &ctx->streams[pkt.stream_index]->codec;
}
- }
- } else {
- AVCodecContext *codec;
- send_it:
- /* Fudge here */
- codec = &c->fmt_ctx.streams[pkt.stream_index]->codec;
-
- codec->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
-
+
+ codec->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); */
+ 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();
+ } else {
+ ret = url_open_dyn_buf(&ctx->pb);
+ }
+ if (ret < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ if (av_write_packet(ctx, &pkt, pkt.pts)) {
+ c->state = HTTPSTATE_SEND_DATA_TRAILER;
+ }
+
+ len = url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
+ codec->frame_number++;
}
+#ifndef AV_READ_FRAME
+ av_free_packet(&pkt);
#endif
-
- if (av_write_packet(&c->fmt_ctx, &pkt, 0))
- c->state = HTTPSTATE_SEND_DATA_TRAILER;
-
- codec->frame_number++;
}
-
- av_free_packet(&pkt);
}
}
break;
default:
case HTTPSTATE_SEND_DATA_TRAILER:
/* last packet test ? */
- if (c->last_packet_sent)
+ if (c->last_packet_sent || c->is_packetized)
return -1;
+ ctx = &c->fmt_ctx;
/* prepare header */
- av_write_trailer(&c->fmt_ctx);
+ if (url_open_dyn_buf(&ctx->pb) < 0) {
+ /* XXX: potential leak */
+ return -1;
+ }
+ av_write_trailer(ctx);
+ len = url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+
c->last_packet_sent = 1;
break;
}
return 0;
}
+/* in bit/s */
+#define SHORT_TERM_BANDWIDTH 8000000
+
/* should convert the format at the same time */
static int http_send_data(HTTPContext *c)
{
- int len, ret;
-
+ 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;
@@ -1684,20 +2133,58 @@ static int http_send_data(HTTPContext *c)
}
}
- if (c->buffer_end > c->buffer_ptr) {
- 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;
+ 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;
}
- } else {
+
+ c->buffer_ptr += 4;
+ url_write(c->rtp_handles[c->packet_stream_index],
+ c->buffer_ptr, len);
c->buffer_ptr += len;
- c->data_count += len;
- update_datarate(&c->datarate, c->data_count);
- if (c->stream)
- c->stream->bytes_served += 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 {
+ 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;
}
return 0;
}
@@ -1827,6 +2314,635 @@ static int http_receive_data(HTTPContext *c)
return -1;
}
+/********************************************************************/
+/* RTSP handling */
+
+static void rtsp_reply_header(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ const char *str;
+ time_t ti;
+ char *p;
+ char buf2[32];
+
+ switch(error_number) {
+#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);
+
+ /* output GMT time */
+ ti = time(NULL);
+ p = ctime(&ti);
+ strcpy(buf2, p);
+ p = buf2 + strlen(p) - 1;
+ if (*p == '\n')
+ *p = '\0';
+ url_fprintf(c->pb, "Date: %s GMT\r\n", buf2);
+}
+
+static void rtsp_reply_error(HTTPContext *c, enum RTSPStatusCode error_number)
+{
+ rtsp_reply_header(c, error_number);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static int rtsp_parse_request(HTTPContext *c)
+{
+ const char *p, *p1, *p2;
+ char cmd[32];
+ char url[1024];
+ char protocol[32];
+ char line[1024];
+ 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);
+
+ pstrcpy(c->method, sizeof(c->method), cmd);
+ pstrcpy(c->url, sizeof(c->url), url);
+ pstrcpy(c->protocol, sizeof(c->protocol), protocol);
+
+ c->pb = &pb1;
+ if (url_open_dyn_buf(c->pb) < 0) {
+ /* XXX: cannot do more */
+ c->pb = NULL; /* safety */
+ return -1;
+ }
+
+ /* check version name */
+ if (strcmp(protocol, "RTSP/1.0") != 0) {
+ rtsp_reply_error(c, RTSP_STATUS_VERSION);
+ goto the_end;
+ }
+
+ /* parse each header line */
+ memset(header, 0, sizeof(RTSPHeader));
+ /* skip to next line */
+ while (*p != '\n' && *p != '\0')
+ p++;
+ if (*p == '\n')
+ p++;
+ while (*p != '\0') {
+ p1 = strchr(p, '\n');
+ if (!p1)
+ break;
+ p2 = p1;
+ if (p2 > p && p2[-1] == '\r')
+ p2--;
+ /* skip empty line */
+ if (p2 == p)
+ break;
+ len = p2 - p;
+ if (len > sizeof(line) - 1)
+ len = sizeof(line) - 1;
+ memcpy(line, p, len);
+ line[len] = '\0';
+ rtsp_parse_line(header, line);
+ p = p1 + 1;
+ }
+
+ /* handle sequence number */
+ c->seq = header->seq;
+
+ if (!strcmp(cmd, "DESCRIBE")) {
+ rtsp_cmd_describe(c, url);
+ } else if (!strcmp(cmd, "SETUP")) {
+ rtsp_cmd_setup(c, url, header);
+ } else if (!strcmp(cmd, "PLAY")) {
+ rtsp_cmd_play(c, url, header);
+ } else if (!strcmp(cmd, "PAUSE")) {
+ rtsp_cmd_pause(c, url, header);
+ } else if (!strcmp(cmd, "TEARDOWN")) {
+ rtsp_cmd_teardown(c, url, header);
+ } else {
+ rtsp_reply_error(c, RTSP_STATUS_METHOD);
+ }
+ the_end:
+ len = url_close_dyn_buf(c->pb, &c->pb_buffer);
+ c->pb = NULL; /* safety */
+ if (len < 0) {
+ /* XXX: cannot do more */
+ return -1;
+ }
+ c->buffer_ptr = c->pb_buffer;
+ c->buffer_end = c->pb_buffer + len;
+ c->state = RTSPSTATE_SEND_REPLY;
+ return 0;
+}
+
+static int prepare_sdp_description(HTTPContext *c,
+ FFStream *stream, UINT8 **pbuffer)
+{
+ ByteIOContext pb1, *pb = &pb1;
+ struct sockaddr_in my_addr;
+ int len, i, payload_type;
+ const char *ipstr, *title, *mediatype;
+ AVStream *st;
+
+ len = sizeof(my_addr);
+ getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
+ ipstr = inet_ntoa(my_addr.sin_addr);
+
+ if (url_open_dyn_buf(pb) < 0)
+ return -1;
+
+ /* general media info */
+
+ url_fprintf(pb, "v=0\n");
+ url_fprintf(pb, "o=- 0 0 IN IP4 %s\n", ipstr);
+ title = stream->title;
+ if (title[0] == '\0')
+ title = "No Title";
+ url_fprintf(pb, "s=%s\n", title);
+ if (stream->comment[0] != '\0')
+ url_fprintf(pb, "i=%s\n", stream->comment);
+
+ /* for each stream, we output the necessary info */
+ 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:
+ mediatype = "video";
+ break;
+ default:
+ mediatype = "application";
+ break;
+ }
+ /* XXX: the port indication is not correct (but should be correct
+ for broadcast) */
+ payload_type = rtp_get_payload_type(&st->codec);
+
+ url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
+ mediatype, 0, payload_type);
+ url_fprintf(pb, "a=control:streamid=%d\n", i);
+ }
+ return url_close_dyn_buf(pb, pbuffer);
+}
+
+static void rtsp_cmd_describe(HTTPContext *c, const char *url)
+{
+ FFStream *stream;
+ char path1[1024];
+ const char *path;
+ UINT8 *content;
+ int content_length;
+
+ /* find which url is asked */
+ url_split(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 &&
+ !strcmp(path, stream->filename)) {
+ goto found;
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
+ return;
+
+ found:
+ /* prepare the media description in sdp format */
+ content_length = prepare_sdp_description(c, stream, &content);
+ if (content_length < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
+ return;
+ }
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ url_fprintf(c->pb, "Content-Type: application/sdp\r\n");
+ url_fprintf(c->pb, "Content-Length: %d\r\n", content_length);
+ url_fprintf(c->pb, "\r\n");
+ put_buffer(c->pb, content, content_length);
+}
+
+static HTTPContext *find_rtp_session(const char *session_id)
+{
+ HTTPContext *c;
+
+ if (session_id[0] == '\0')
+ return NULL;
+
+ for(c = first_http_ctx; c != NULL; c = c->next) {
+ if (!strcmp(c->session_id, session_id))
+ return c;
+ }
+ return NULL;
+}
+
+RTSPTransportField *find_transport(RTSPHeader *h, enum RTSPProtocol protocol)
+{
+ RTSPTransportField *th;
+ int i;
+
+ for(i=0;i<h->nb_transports;i++) {
+ th = &h->transports[i];
+ if (th->protocol == protocol)
+ return th;
+ }
+ return NULL;
+}
+
+static void rtsp_cmd_setup(HTTPContext *c, const char *url,
+ RTSPHeader *h)
+{
+ FFStream *stream;
+ int stream_index, port;
+ char buf[1024];
+ char path1[1024];
+ const char *path;
+ HTTPContext *rtp_c;
+ 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);
+ 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) {
+ /* accept aggregate filenames only if single stream */
+ if (!strcmp(path, stream->filename)) {
+ if (stream->nb_streams != 1) {
+ rtsp_reply_error(c, RTSP_STATUS_AGGREGATE);
+ return;
+ }
+ stream_index = 0;
+ goto found;
+ }
+
+ for(stream_index = 0; stream_index < stream->nb_streams;
+ stream_index++) {
+ snprintf(buf, sizeof(buf), "%s/streamid=%d",
+ stream->filename, stream_index);
+ if (!strcmp(path, buf))
+ goto found;
+ }
+ }
+ }
+ /* no stream found */
+ rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
+ return;
+ found:
+
+ /* generate session id if needed */
+ if (h->session_id[0] == '\0') {
+ 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, stream, h->session_id);
+ if (!rtp_c) {
+ rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
+ return;
+ }
+
+ /* open input stream */
+ if (open_input_stream(rtp_c, "") < 0) {
+ 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);
+ return;
+ }
+
+ /* check transport */
+ th = find_transport(h, rtp_c->rtp_protocol);
+ if (!th || (th->protocol == RTSP_PROTOCOL_RTP_UDP &&
+ th->client_port_min <= 0)) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* setup default options */
+ 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,
+ (char *)&setup, sizeof(setup),
+ stream->rtsp_option) < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+ dest_addr.sin_addr.s_addr = htonl(setup.ipaddr);
+ }
+
+ /* setup stream */
+ if (rtp_new_av_stream(rtp_c, stream_index, &dest_addr) < 0) {
+ rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
+ return;
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+
+ switch(rtp_c->rtp_protocol) {
+ case RTSP_PROTOCOL_RTP_UDP:
+ port = rtp_get_local_port(rtp_c->rtp_handles[stream_index]);
+ url_fprintf(c->pb, "Transport: RTP/AVP/UDP;unicast;"
+ "client_port=%d-%d;server_port=%d-%d",
+ th->client_port_min, th->client_port_min + 1,
+ port, port + 1);
+ break;
+ case RTSP_PROTOCOL_RTP_TCP:
+ url_fprintf(c->pb, "Transport: RTP/AVP/TCP;interleaved=%d-%d",
+ stream_index * 2, stream_index * 2 + 1);
+ break;
+ default:
+ break;
+ }
+ if (setup.transport_option[0] != '\0') {
+ 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,
+ const char *session_id)
+{
+ HTTPContext *rtp_c;
+ char path1[1024];
+ const char *path;
+
+ 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);
+ path = path1;
+ if (*path == '/')
+ path++;
+ if (strcmp(path, rtp_c->stream->filename) != 0)
+ return NULL;
+ return rtp_c;
+}
+
+static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ 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) {
+ rtsp_reply_error(c, RTSP_STATUS_STATE);
+ return;
+ }
+
+ 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 */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ 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;
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPHeader *h)
+{
+ HTTPContext *rtp_c;
+
+ rtp_c = find_rtp_session_with_url(url, h->session_id);
+ if (!rtp_c) {
+ 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,
+ NULL, 0,
+ rtp_c->stream->rtsp_option);
+ }
+
+ /* now everything is OK, so we can send the connection parameters */
+ rtsp_reply_header(c, RTSP_STATUS_OK);
+ /* session ID */
+ url_fprintf(c->pb, "Session: %s\r\n", rtp_c->session_id);
+ url_fprintf(c->pb, "\r\n");
+}
+
+
+/********************************************************************/
+/* RTP handling */
+
+static HTTPContext *rtp_new_connection(HTTPContext *rtsp_c,
+ FFStream *stream, const char *session_id)
+{
+ HTTPContext *c = NULL;
+
+ /* 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 = rtsp_c->from_addr;
+ c->buffer_size = IOBUFFER_INIT_SIZE;
+ c->buffer = av_malloc(c->buffer_size);
+ if (!c->buffer)
+ goto fail;
+ nb_connections++;
+ c->stream = stream;
+ pstrcpy(c->session_id, sizeof(c->session_id), session_id);
+ c->state = HTTPSTATE_READY;
+ c->is_packetized = 1;
+ /* protocol is shown in statistics */
+ pstrcpy(c->protocol, sizeof(c->protocol), "RTP");
+
+ c->next = first_http_ctx;
+ first_http_ctx = c;
+ return c;
+
+ fail:
+ if (c) {
+ av_free(c->buffer);
+ av_free(c);
+ }
+ return NULL;
+}
+
+/* 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
+ used. */
+static int rtp_new_av_stream(HTTPContext *c,
+ int stream_index, struct sockaddr_in *dest_addr)
+{
+ AVFormatContext *ctx;
+ AVStream *st;
+ char *ipaddr;
+ URLContext *h;
+ UINT8 *dummy_buf;
+
+ /* now we can open the relevant output stream */
+ ctx = av_mallocz(sizeof(AVFormatContext));
+ if (!ctx)
+ return -1;
+ ctx->oformat = &rtp_mux;
+
+ st = av_mallocz(sizeof(AVStream));
+ if (!st)
+ goto fail;
+ ctx->nb_streams = 1;
+ ctx->streams[0] = st;
+
+ if (!c->stream->feed ||
+ c->stream->feed == c->stream) {
+ memcpy(st, c->stream->streams[stream_index], sizeof(AVStream));
+ } else {
+ 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);
+
+ snprintf(ctx->filename, sizeof(ctx->filename),
+ "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
+
+ printf("open %s\n", ctx->filename);
+
+ if (url_open(&h, ctx->filename, URL_WRONLY) < 0)
+ goto fail;
+ c->rtp_handles[stream_index] = h;
+ } else {
+ goto fail;
+ }
+
+ /* 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) {
+ /* XXX: close stream */
+ goto fail;
+ }
+ if (av_write_header(ctx) < 0) {
+ fail:
+ if (h)
+ url_close(h);
+ av_free(ctx);
+ return -1;
+ }
+ url_close_dyn_buf(&ctx->pb, &dummy_buf);
+ av_free(dummy_buf);
+
+ c->rtp_ctx[stream_index] = ctx;
+ return 0;
+}
+
+/********************************************************************/
+/* ffserver initialization */
+
+AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec)
+{
+ AVStream *fst;
+
+ fst = av_mallocz(sizeof(AVStream));
+ if (!fst)
+ return NULL;
+ fst->priv_data = av_mallocz(sizeof(FeedData));
+ memcpy(&fst->codec, codec, sizeof(AVCodecContext));
+ stream->streams[stream->nb_streams++] = fst;
+ return fst;
+}
+
/* return the stream number in the feed */
int add_av_stream(FFStream *feed,
AVStream *st)
@@ -1862,17 +2978,66 @@ int add_av_stream(FFStream *feed,
}
}
- fst = av_mallocz(sizeof(AVStream));
+ fst = add_av_stream1(feed, av);
if (!fst)
return -1;
- fst->priv_data = av_mallocz(sizeof(FeedData));
- memcpy(&fst->codec, av, sizeof(AVCodecContext));
- feed->streams[feed->nb_streams++] = fst;
return feed->nb_streams - 1;
found:
return i;
}
+void remove_stream(FFStream *stream)
+{
+ FFStream **ps;
+ ps = &first_stream;
+ while (*ps != NULL) {
+ if (*ps == stream) {
+ *ps = (*ps)->next;
+ } else {
+ ps = &(*ps)->next;
+ }
+ }
+}
+
+/* compute the needed AVStream for each file */
+void build_file_streams(void)
+{
+ FFStream *stream, *stream_next;
+ AVFormatContext *infile;
+ int i;
+
+ /* gather all streams */
+ for(stream = first_stream; stream != NULL; stream = stream_next) {
+ stream_next = stream->next;
+ if (stream->stream_type == STREAM_TYPE_LIVE &&
+ !stream->feed) {
+ /* 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) {
+ http_log("%s not found", stream->feed_filename);
+ /* remove stream (no need to spend more time on it) */
+ fail:
+ remove_stream(stream);
+ } else {
+ /* 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'",
+ stream->feed_filename);
+ av_close_input_file(infile);
+ goto fail;
+ }
+ for(i=0;i<infile->nb_streams;i++) {
+ add_av_stream1(stream, &infile->streams[i]->codec);
+ }
+ av_close_input_file(infile);
+ }
+ }
+ }
+}
+
/* compute the needed AVStream for each feed */
void build_feed_streams(void)
{
@@ -1884,6 +3049,7 @@ void build_feed_streams(void)
feed = stream->feed;
if (feed) {
if (!stream->is_feed) {
+ /* we handle a stream coming from a feed */
for(i=0;i<stream->nb_streams;i++) {
stream->feed_streams[i] = add_av_stream(feed, stream->streams[i]);
}
@@ -2060,6 +3226,30 @@ int opt_video_codec(const char *arg)
return p->id;
}
+/* simplistic plugin support */
+
+void load_module(const char *filename)
+{
+ void *dll;
+ void (*init_func)(void);
+ dll = dlopen(filename, RTLD_NOW);
+ if (!dll) {
+ fprintf(stderr, "Could not load module '%s' - %s\n",
+ filename, dlerror());
+ return;
+ }
+
+ init_func = dlsym(dll, "ffserver_module_init");
+ if (!init_func) {
+ fprintf(stderr,
+ "%s: init function 'ffserver_module_init()' not found\n",
+ filename);
+ dlclose(dll);
+ }
+
+ init_func();
+}
+
int parse_ffconfig(const char *filename)
{
FILE *f;
@@ -2104,10 +3294,22 @@ int parse_ffconfig(const char *filename)
if (!strcasecmp(cmd, "Port")) {
get_arg(arg, sizeof(arg), &p);
- my_addr.sin_port = htons (atoi(arg));
+ my_http_addr.sin_port = htons (atoi(arg));
} else if (!strcasecmp(cmd, "BindAddress")) {
get_arg(arg, sizeof(arg), &p);
- if (!inet_aton(arg, &my_addr.sin_addr)) {
+ if (!inet_aton(arg, &my_http_addr.sin_addr)) {
+ fprintf(stderr, "%s:%d: Invalid IP address: %s\n",
+ filename, line_num, arg);
+ errors++;
+ }
+ } else if (!strcasecmp(cmd, "NoDaemon")) {
+ ffserver_daemon = 0;
+ } else if (!strcasecmp(cmd, "RTSPPort")) {
+ get_arg(arg, sizeof(arg), &p);
+ my_rtsp_addr.sin_port = htons (atoi(arg));
+ } 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",
filename, line_num, arg);
errors++;
@@ -2185,7 +3387,7 @@ int parse_ffconfig(const char *filename)
feed->child_argv[i] = av_malloc(30 + strlen(feed->filename));
snprintf(feed->child_argv[i], 256, "http://127.0.0.1:%d/%s",
- ntohs(my_addr.sin_port), feed->filename);
+ ntohs(my_http_addr.sin_port), feed->filename);
}
} else if (!strcasecmp(cmd, "File")) {
if (feed) {
@@ -2428,6 +3630,16 @@ int parse_ffconfig(const char *filename)
video_id = CODEC_ID_NONE;
} else if (!strcasecmp(cmd, "NoAudio")) {
audio_id = CODEC_ID_NONE;
+ } else if (!strcasecmp(cmd, "RTSPOption")) {
+ get_arg(arg, sizeof(arg), &p);
+ if (stream) {
+ av_freep(&stream->rtsp_option);
+ /* XXX: av_strdup ? */
+ stream->rtsp_option = av_malloc(strlen(arg) + 1);
+ if (stream->rtsp_option) {
+ strcpy(stream->rtsp_option, arg);
+ }
+ }
} else if (!strcasecmp(cmd, "</Stream>")) {
if (!stream) {
fprintf(stderr, "%s:%d: No corresponding <Stream> for </Stream>\n",
@@ -2481,6 +3693,9 @@ int parse_ffconfig(const char *filename)
errors++;
}
redirect = NULL;
+ } else if (!strcasecmp(cmd, "LoadModule")) {
+ get_arg(arg, sizeof(arg), &p);
+ load_module(arg);
} else {
fprintf(stderr, "%s:%d: Incorrect keyword: '%s'\n",
filename, line_num, cmd);
@@ -2496,12 +3711,6 @@ int parse_ffconfig(const char *filename)
}
-void *http_server_thread(void *arg)
-{
- http_server(my_addr);
- return NULL;
-}
-
#if 0
static void write_packet(FFCodec *ffenc,
UINT8 *buf, int size)
@@ -2590,9 +3799,10 @@ int main(int argc, char **argv)
config_filename = "/etc/ffserver.conf";
my_program_name = argv[0];
-
+ ffserver_daemon = 1;
+
for(;;) {
- c = getopt_long_only(argc, argv, "ndLh?f:", NULL, NULL);
+ c = getopt(argc, argv, "ndLh?f:");
if (c == -1)
break;
switch(c) {
@@ -2608,6 +3818,7 @@ int main(int argc, char **argv)
break;
case 'd':
ffserver_debug = 1;
+ ffserver_daemon = 0;
break;
case 'f':
config_filename = optarg;
@@ -2619,10 +3830,16 @@ int main(int argc, char **argv)
putenv("http_proxy"); /* Kill the http_proxy */
- /* address on which the server will handle connections */
- my_addr.sin_family = AF_INET;
- my_addr.sin_port = htons (8080);
- my_addr.sin_addr.s_addr = htonl (INADDR_ANY);
+ /* address on which the server will handle HTTP connections */
+ my_http_addr.sin_family = AF_INET;
+ my_http_addr.sin_port = htons (8080);
+ my_http_addr.sin_addr.s_addr = htonl (INADDR_ANY);
+
+ /* address on which the server will handle RTSP connections */
+ 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;
nb_max_bandwidth = 1000;
first_stream = NULL;
@@ -2638,8 +3855,34 @@ int main(int argc, char **argv)
exit(1);
}
+ build_file_streams();
+
build_feed_streams();
+ /* put the process in background and detach it from its TTY */
+ if (ffserver_daemon) {
+ int pid;
+
+ pid = fork();
+ if (pid < 0) {
+ perror("fork");
+ exit(1);
+ } else if (pid > 0) {
+ /* parent : exit */
+ exit(0);
+ } else {
+ /* child */
+ setsid();
+ chdir("/");
+ close(0);
+ close(1);
+ close(2);
+ open("/dev/null", O_RDWR);
+ dup(0);
+ dup(0);
+ }
+ }
+
/* signal init */
signal(SIGPIPE, SIG_IGN);
@@ -2651,8 +3894,8 @@ int main(int argc, char **argv)
logfile = fopen(logfilename, "w");
}
- if (http_server(my_addr) < 0) {
- fprintf(stderr, "Could not start http server\n");
+ if (http_server() < 0) {
+ fprintf(stderr, "Could not start server\n");
exit(1);
}
--
Libav/FFmpeg packaging
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